Grandstream Networks, Inc.

Grandstream Networks, Inc.

UCM6100 Series IP PBX

User Manual

COPYRIGHT

©2017 Grandstream Networks, Inc. http://www.grandstream.com

All rights reserved. Information in this document is subject to change without notice. Reproduction or transmittal of the entire or any part, in any form or by any means, electronic or print, for any purpose without the express written permission of Grandstream Networks, Inc. is not permitted.

The latest electronic version of this user manual is available for download here: http://www.grandstream.com/support

Grandstream is a registered trademark and Grandstream logo is trademark of Grandstream Networks, Inc. in the United States, Europe and other countries.

CAUTION

Changes or modifications to this product not expressly approved by Grandstream, or operation of this product in any way other than as detailed by this User Manual, could void your manufacturer warranty.

WARNING

Please do not use a different power adaptor with your devices as it may cause damage to the products and void the manufacturer warranty.

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UCM6100 Series User Manual

Version 1.0.15.16

GNU GPL INFORMATION

UCM6100 firmware contains third-party software licensed under the GNU General Public License (GPL).

Grandstream uses software under the specific terms of the GPL. Please see the GNU General Public

License (GPL) for the exact terms and conditions of the license.

Grandstream GNU GPL related source code can be downloaded from Grandstream web site from: http://www.grandstream.com/support/faq/gnu-general-public-license/gnu-gpl-information-download

UCM6100 Series User Manual

Version 1.0.15.16

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Table of Content

DOCUMENT PURPOSE ........................................................................... 22

CHANGE LOG ......................................................................................... 23

Firmware Version 1.0.15.16 ................................................................................................................. 23

Firmware Version 1.0.14.24 ................................................................................................................. 23

Firmware Version 1.0.14.23 ................................................................................................................. 23

Firmware Version 1.0.14.21 ................................................................................................................. 24

Firmware Version 1.0.13.14 ................................................................................................................. 24

Firmware Version 1.0.12.19 ................................................................................................................. 25

Firmware Version 1.0.11.27 ................................................................................................................. 26

Firmware Version 1.0.10.44 ................................................................................................................. 26

Firmware Version 1.0.10.39 ................................................................................................................. 26

Firmware Version 1.0.9.26 ................................................................................................................... 27

Firmware Version 1.0.9.25 ................................................................................................................... 27

Firmware Version 1.0.9.23 ................................................................................................................... 27

Firmware Version 1.0.8.12 ................................................................................................................... 28

Firmware Version 1.0.7.11 ................................................................................................................... 29

Firmware Version 1.0.6.11 ................................................................................................................... 30

Firmware Version 1.0.6.10 ................................................................................................................... 30

Firmware Version 1.0.5.19 ................................................................................................................... 30

Firmware Version 1.0.5.14 ................................................................................................................... 30

Firmware Version 1.0.4.7 ..................................................................................................................... 31

Firmware Version 1.0.3.13 ................................................................................................................... 31

Firmware Version 1.0.2.21 ................................................................................................................... 31

Firmware Version 1.0.1.22 ................................................................................................................... 32

WELCOME ............................................................................................... 33

PRODUCT OVERVIEW ............................................................................ 34

Technical Specifications ...................................................................................................................... 34

INSTALLATION ........................................................................................ 37

Equipment Packaging .......................................................................................................................... 37

Connect Your UCM6100 ...................................................................................................................... 37

Connect The UCM6102 ................................................................................................................ 37

Connect The UCM6104 ................................................................................................................ 38

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Connect The UCM6108 ................................................................................................................ 39

Connect The UCM6116 ................................................................................................................ 40

Safety Compliances ............................................................................................................................. 41

Warranty .............................................................................................................................................. 41

GETTING STARTED ................................................................................ 42

Use The LCD Menu ............................................................................................................................. 42

Use The LED Indicators ...................................................................................................................... 44

Use The Web GUI ............................................................................................................................... 45

Access Web GUI .......................................................................................................................... 45

Setup Wizard ................................................................................................................................ 46

Web GUI Configurations ............................................................................................................... 47

Web GUI Languages .................................................................................................................... 47

Save and Apply Changes ............................................................................................................. 48

Make your First Call ............................................................................................................................. 48

SYSTEM SETTINGS ................................................................................ 49

Network Settings ................................................................................................................................. 49

Basic Settings ............................................................................................................................... 49

DHCP Client List ........................................................................................................................... 57

802.1X .......................................................................................................................................... 58

Static Routes ................................................................................................................................ 60

Port Forwarding (UCM6102 only) ................................................................................................. 62

OpenVPN...................................................................................................................................... 64

DDNS Settings ............................................................................................................................. 66

Security Settings .................................................................................................................................. 68

Static Defense .............................................................................................................................. 68

Dynamic Defense ......................................................................................................................... 71

Fail2ban ........................................................................................................................................ 72

SSH Access .................................................................................................................................. 74

LDAP Server ........................................................................................................................................ 75

LDAP Server Configurations ........................................................................................................ 76

LDAP Phonebook ......................................................................................................................... 77

LDAP Client Configurations .......................................................................................................... 80

HTTP Server ........................................................................................................................................ 82

Email .................................................................................................................................................... 83

Email Settings ............................................................................................................................... 83

Email Templates ........................................................................................................................... 85

Email Send Log ............................................................................................................................ 86

Time Settings ....................................................................................................................................... 88

Auto Time Updating ...................................................................................................................... 88

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Set Time Manually ........................................................................................................................ 89

NTP Server ................................................................................................................................... 90

Office Time ................................................................................................................................... 90

Holiday .......................................................................................................................................... 91

Recordings Storage ............................................................................................................................. 93

Google Service Settings Support ........................................................................................................ 95

PROVISIONING ....................................................................................... 98

Overview .............................................................................................................................................. 98

Configuration Architecture for End Point Device ................................................................................. 98

Auto Provisioning Settings ................................................................................................................... 99

Discovery ........................................................................................................................................... 102

Uploading Devices List ...................................................................................................................... 104

Managing discovered devices: .......................................................................................................... 104

Global Configuration .......................................................................................................................... 105

Global Policy ............................................................................................................................... 105

Global Templates ........................................................................................................................ 113

Model Configuration .......................................................................................................................... 114

Model Templates ........................................................................................................................ 114

Model Update ............................................................................................................................. 116

Device Configuration ......................................................................................................................... 118

Create New Device ..................................................................................................................... 118

Manage Devices ......................................................................................................................... 119

Sample Application ............................................................................................................................ 126

EXTENSIONS ......................................................................................... 131

Create New User ............................................................................................................................... 131

Create New SIP Extension ......................................................................................................... 131

Create New IAX Extension ......................................................................................................... 139

Create New FXS Extension ........................................................................................................ 144

Batch Add Extensions ........................................................................................................................ 149

Batch Add SIP Extensions .......................................................................................................... 149

Batch Add IAX Extensions .......................................................................................................... 152

Search and Edit Extension ................................................................................................................ 154

Export Extensions .............................................................................................................................. 156

Import Extensions .............................................................................................................................. 156

E-mail Notification .............................................................................................................................. 163

Multiple Registrations per Extension ................................................................................................. 165

SMS Message Support ...................................................................................................................... 166

EXTENSION GROUPS ........................................................................... 167

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Configure Extension Groups ............................................................................................................. 167

Using Extension Groups .................................................................................................................... 168

ANALOG TRUNKS ................................................................................ 169

Analog Trunk Configuration ............................................................................................................... 169

PSTN Detection ................................................................................................................................. 172

VOIP TRUNKS ....................................................................................... 175

Direct Outward Dialing (DOD) ........................................................................................................... 185

SLA STATION ........................................................................................ 187

Create/Edit SLA Station ..................................................................................................................... 187

Sample Configuration ........................................................................................................................ 188

CALL ROUTES ...................................................................................... 190

Outbound Routes .............................................................................................................................. 190

Configuring Outbound Routes .................................................................................................... 190

Outbound Blacklist ...................................................................................................................... 193

PIN Groups ................................................................................................................................. 194

Inbound Routes ................................................................................................................................. 197

Inbound Rule Configurations ...................................................................................................... 198

Inbound Route: Prepend Example ............................................................................................. 200

Inbound Route: Multiple Mode ................................................................................................... 201

Fax Intelligent Route ................................................................................................................... 202

Fax with Two Media .................................................................................................................... 202

Blacklist Configurations .............................................................................................................. 203

CONFERENCE ROOM ........................................................................... 205

Conference Room Configurations .............................................................................................. 205

Join a Conference Call ............................................................................................................... 207

Invite Other Parties to Join Conference ..................................................................................... 207

During The Conference .............................................................................................................. 208

Record Conference .................................................................................................................... 210

CONFERENCE SCHEDULE .................................................................. 211

Conference Schedule Configuration ................................................................................................. 211

IVR ......................................................................................................... 215

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Configure IVR .................................................................................................................................... 215

Black/White List in IVR ...................................................................................................................... 218

Create Custom Prompt ...................................................................................................................... 220

LANGUAGE SETTINGS FOR VOICE PROMPT .................................... 221

Download and Install Voice Prompt Package .................................................................................... 221

Customize Specific Prompt ............................................................................................................... 223

VOICEMAIL ............................................................................................ 224

Configure Voicemail ........................................................................................................................... 224

Access Voicemail ............................................................................................................................... 226

Extension Voicemail Count ................................................................................................................ 227

Voicemail Email Settings ................................................................................................................... 227

Configure Voicemail Group ................................................................................................................ 228

RING GROUP ......................................................................................... 230

Configure Ring Group ........................................................................................................................ 230

Remote Extension in Ring Group ...................................................................................................... 232

PAGING AND INTERCOM GROUP ....................................................... 235

Configure Paging/Intercom Group ..................................................................................................... 235

CALL QUEUE ........................................................................................ 237

Configure Call Queue ........................................................................................................................ 237

Call Center Settings & Enhancement ................................................................................................ 241

Queue Statistics ................................................................................................................................ 243

Switchboard ....................................................................................................................................... 244

PICKUP GROUPS .................................................................................. 246

Configure Pickup Groups .................................................................................................................. 246

Configure Pickup Feature Code ........................................................................................................ 246

MUSIC ON HOLD ................................................................................... 248

FAX SERVER ......................................................................................... 251

Configure Fax/T.38 ............................................................................................................................ 251

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Receiving Fax .................................................................................................................................... 253

Sample Configuration to Receive Fax from PSTN Line ............................................................. 253

Sample Configuration for Fax-To-Email ..................................................................................... 255

Fax Sending....................................................................................................................................... 256

BUSY CAMP-ON .................................................................................... 258

PRESENCE ............................................................................................ 259

FOLLOW ME .......................................................................................... 261

SPEED DIAL .......................................................................................... 263

DISA ....................................................................................................... 264

CALLBACK ............................................................................................ 266

BLF AND EVENT LIST ........................................................................... 267

BLF .................................................................................................................................................... 267

Event List ........................................................................................................................................... 267

DIAL BY NAME ...................................................................................... 270

Dial By Name Configuration .............................................................................................................. 270

User Name Prompt Customization .................................................................................................... 273

Upload User Name Prompt File from Web GUI ......................................................................... 273

Record User Name via Voicemail Menu ..................................................................................... 273

ACTIVE CALLS AND MONITOR ........................................................... 274

Active Calls Status ............................................................................................................................. 274

Hang Up Active Calls ......................................................................................................................... 276

Call Monitor........................................................................................................................................ 276

CALL FEATURES .................................................................................. 278

Feature Codes ................................................................................................................................... 278

Call Recording ................................................................................................................................... 283

Call Park ............................................................................................................................................ 284

Park a Call .................................................................................................................................. 284

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Retrieve Parked Call ................................................................................................................... 284

Enable Spy ........................................................................................................................................ 285

PBX SETTINGS ..................................................................................... 286

General Settings ................................................................................................................................ 286

Voice Prompts Customization............................................................................................................ 288

Record New Custom Prompt ...................................................................................................... 288

Upload Custom Prompt .............................................................................................................. 289

Download All Custom Prompt ..................................................................................................... 289

PBX Settings/Jitter Buffer .................................................................................................................. 290

PBX Settings/RTP Settings ............................................................................................................... 290

PBX Settings/Payload ....................................................................................................................... 291

IAX SETTINGS ....................................................................................... 292

IAX Settings/General ......................................................................................................................... 292

IAX Settings/Registration ................................................................................................................... 292

IAX Settings/Security ......................................................................................................................... 293

SIP SETTINGS ....................................................................................... 294

SIP Settings/General ......................................................................................................................... 294

SIP Settings/Misc............................................................................................................................... 294

SIP Settings/Session Timer ............................................................................................................... 295

SIP Settings/TCP and TLS ................................................................................................................ 296

SIP Settings/NAT ............................................................................................................................... 297

SIP Settings/TOS ............................................................................................................................... 297

SIP Settings/SIP Trunk Prompt Tone ................................................................................................. 299

Transparent Call-Info header ............................................................................................................. 300

INTERFACE SETTINGS ......................................................................... 301

CTI SERVER .......................................................................................... 304

ASTERISK MANAGER INTERFACE (RESTRICTED ACCESS) ............ 305

PMS INTEGRATION ............................................................................... 306

HMobile PMS Connector ................................................................................................................... 306

Mitel PMS .......................................................................................................................................... 307

Configuration ..................................................................................................................................... 308

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Basic Settings ............................................................................................................................. 308

PMS Features .................................................................................................................................... 308

Room Status ............................................................................................................................... 308

Wakeup Service.......................................................................................................................... 310

Mini Bar....................................................................................................................................... 311

CRM INTEGRATION .............................................................................. 314

SugarCRM ......................................................................................................................................... 314

Salesforce CRM ................................................................................................................................ 316

WAKEUP SERVICE ............................................................................... 318

WakeUp Service using Admin Login ................................................................................................. 318

WakeUp Service from User portal ..................................................................................................... 319

WakeUp Service using Feature Code ............................................................................................... 319

ANNOUNCEMENTS CENTER ............................................................... 320

Announcements Center Settings ....................................................................................................... 320

Group Settings ................................................................................................................................... 321

STATUS AND REPORTING ................................................................... 323

PBX Status ........................................................................................................................................ 323

Trunks ......................................................................................................................................... 323

Extensions .................................................................................................................................. 325

Interfaces Status ......................................................................................................................... 326

System Status.................................................................................................................................... 327

General ....................................................................................................................................... 327

Network....................................................................................................................................... 328

Storage Usage ............................................................................................................................ 328

Resource Usage ......................................................................................................................... 329

System Events ................................................................................................................................... 330

Alert Events List .......................................................................................................................... 330

Alert Log ..................................................................................................................................... 332

Alert Contact ............................................................................................................................... 333

CDR ................................................................................................................................................... 333

CDR Improvement ...................................................................................................................... 338

Downloaded CDR File ................................................................................................................ 339

Statistics ..................................................................................................................................... 340

Recording Files ........................................................................................................................... 342

API Configuration ....................................................................................................................... 342

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USER PORTAL ...................................................................................... 344

Basic Information ............................................................................................................................... 346

Personal Data .................................................................................................................................... 346

Value-added Features ....................................................................................................................... 346

MAINTENANCE ..................................................................................... 347

User Management ............................................................................................................................. 347

User Information ......................................................................................................................... 347

Custom Privilege ........................................................................................................................ 348

Concurrent Multi-User Login....................................................................................................... 350

Operation Log ............................................................................................................................. 350

Change Password ...................................................................................................................... 352

Change Binding Email ................................................................................................................ 353

Login Settings ............................................................................................................................. 354

Upgrading .......................................................................................................................................... 355

Upgrading via Network ............................................................................................................... 355

Upgrading via Local Upload ....................................................................................................... 357

No Local Firmware Servers ........................................................................................................ 358

Backup ............................................................................................................................................... 359

Backup/Restore .......................................................................................................................... 359

Data Sync ................................................................................................................................... 361

Restore Configuration from Backup File .................................................................................... 363

System Cleanup/Reset ...................................................................................................................... 364

Reset and Reboot ....................................................................................................................... 364

Cleaner ....................................................................................................................................... 364

USB/SD Card Files Cleanup ...................................................................................................... 366

Syslog ................................................................................................................................................ 367

Network Troubleshooting ................................................................................................................... 367

Ethernet Capture ........................................................................................................................ 367

IP Ping ........................................................................................................................................ 368

Traceroute ................................................................................................................................... 369

Signaling Troubleshooting ................................................................................................................. 370

Analog Record Trace .................................................................................................................. 370

Service Check ............................................................................................................................ 371

Network Status ........................................................................................................................... 372

EXPERIENCING THE UCM6100 SERIES IP PBX ................................. 373

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Table of Tables

Table 1: Technical Specifications ................................................................................................................ 34

Table 2: UCM6102/UCM6104 Equipment Packaging ................................................................................. 37

Table 3: LCD Menu Options ........................................................................................................................ 43

Table 4: UCM6102/UCM6104 LED Indicators ............................................................................................ 44

Table 5: UCM6108/UCM6116 LED Indicators ............................................................................................. 44

Table 6: UCM6102 Network SettingsBasic Settings................................................................................ 49

Table 7: UCM6104 Network SettingsBasic Settings................................................................................ 53

Table 8: UCM6108/UCM6116 Network SettingsBasic Settings .............................................................. 54

Table 9: UCM6100 Network Settings802.1X ........................................................................................... 60

Table 10: UCM6100 Network SettingsStatic Routes ............................................................................... 60

Table 11: UCM6102 Network SettingsPort Forwarding ........................................................................... 62

Table 12: UCM6100 System SettingsOpenVPN ..................................................................................... 64

Table 13: UCM6100 FirewallStatic DefenseCurrent Service ............................................................... 68

Table 14: Typical Firewall Settings .............................................................................................................. 69

Table 15: Firewall Rule Settings .................................................................................................................. 70

Table 16: UCM6100 Firewall Dynamic Defense ......................................................................................... 71

Table 17: Fail2Ban Settings ........................................................................................................................ 74

Table 18: HTTP Server Settings .................................................................................................................. 83

Table 19: Email Settings .............................................................................................................................. 83

Table 20: Email Log ..................................................................................................................................... 87

Table 21: Time Auto Updating ..................................................................................................................... 88

Table 22: Create New Office Time .............................................................................................................. 91

Table 23: Create New Holiday..................................................................................................................... 92

Table 24: Auto Provision Settings ............................................................................................................. 101

Table 25: Global Policy ParametersLocalization ................................................................................... 106

Table 26: Global Policy ParametersPhone Settings .............................................................................. 107

Table 27: Global Policy ParametersContact List ................................................................................... 108

Table 28: Global Policy ParametersMaintenance.................................................................................. 109

Table 29: Global Policy ParametersNetwork Settings ........................................................................... 111

Table 30: Global Policy ParametersCustomization ............................................................................... 112

Table 31: Create New Template ................................................................................................................ 113

Table 32: Create New Model Template ..................................................................................................... 115

Table 33: SIP Extension Configuration ParametersBasic Settings ....................................................... 132

Table 34: SIP Extension Configuration ParametersMedia .................................................................... 133

Table 35: SIP Extension Configuration ParametersFeatures ................................................................ 135

Table 36: SIP Extension Configuration ParametersSpecific Time ......................................................... 139

Table 37: IAX Extension Configuration ParametersBasic Settings ....................................................... 139

Table 38: IAX Extension Configuration ParametersMedia .................................................................... 141

Table 39: IAX Extension Configuration ParametersFeatures ................................................................ 141

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Table 40: IAX Extension Configuration ParametersSpecific Time ......................................................... 144

Table 41: FXS Extension Configuration ParametersBasic Settings ...................................................... 144

Table 42: FXS Extension Configuration ParametersMedia ................................................................... 145

Table 43: FXS Extension Configuration ParametersFeatures ............................................................... 146

Table 44: FXS Extension Configuration ParametersSpecific Time ....................................................... 149

Table 45: Batch Add SIP Extension Parameters ....................................................................................... 149

Table 46: Batch Add IAX Extension Parameters ....................................................................................... 152

Table 47: SIP extensions Imported File Example ..................................................................................... 157

Table 48: IAX extensions Imported File Example ..................................................................................... 160

Table 49: FXS extensions Imported File Example .................................................................................... 161

Table 50: Analog Trunk Configuration Parameters ................................................................................... 169

Table 51: PSTN Detection for Analog Trunk ............................................................................................. 174

Table 52: Create New SIP Trunk ............................................................................................................... 175

Table 53: SIP Register Trunk Configuration Parameters .......................................................................... 176

Table 54: SIP Peer Trunk Configuration Parameters ................................................................................ 180

Table 55: Create New IAX Trunk ............................................................................................................... 182

Table 56: IAX Register Trunk Configuration Parameters .......................................................................... 182

Table 57: IAX Peer Trunk Configuration Parameters ................................................................................ 183

Table 58: SLA Station Configuration Parameters ..................................................................................... 187

Table 59: Outbound Route Configuration Parameters .............................................................................. 190

Table 60: PIN Group .................................................................................................................................. 194

Table 61: Inbound Rule Configuration Parameters ................................................................................... 198

Table 62: Conference Room Configuration Parameters ........................................................................... 205

Table 63: Conference Settings .................................................................................................................. 207

Table 64: Conference Caller IVR Menu .................................................................................................... 209

Table 65: Conference Schedule Parameters ............................................................................................ 211

Table 66: IVR Configuration Parameters .................................................................................................. 216

Table 67: Voicemail Settings ..................................................................................................................... 225

Table 68: Voicemail IVR Menu .................................................................................................................. 226

Table 69: Voicemail Email Settings ........................................................................................................... 227

Table 70: Voicemail Group Settings .......................................................................................................... 229

Table 71: Ring Group Parameters ............................................................................................................ 230

Table 72: Paging/Intercom Group Configuration Parameters ................................................................... 235

Table 73: Call Queue Configuration Parameters ...................................................................................... 237

Table 74: Static Agents limitation .............................................................................................................. 240

Table 75: Call Center Parameters ............................................................................................................. 242

Table 76: Switchboard Parameters ........................................................................................................... 244

Table 77: FAX/T.38 Settings ...................................................................................................................... 252

Table 78: SIP Presence Status ................................................................................................................. 260

Table 79: Follow Me Settings .................................................................................................................... 262

Table 80: Follow Me Options ..................................................................................................................... 262

Table 81: DISA Settings ............................................................................................................................ 265

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Table 82: Callback Configuration Parameters........................................................................................... 266

Table 83: Event List Settings ..................................................................................................................... 267

Table 84: UCM6100 Feature Codes ......................................................................................................... 278

Table 85: Internal Options/General ........................................................................................................... 286

Table 86: Internal Options/Jitter Buffer ...................................................................................................... 290

Table 87: Internal Options/RTP Settings ................................................................................................... 290

Table 88: Internal Options/Payload ........................................................................................................... 291

Table 89: IAX Settings/General ................................................................................................................. 292

Table 90: IAX Settings/Registration .......................................................................................................... 292

Table 91: IAX Settings/Static Defense ...................................................................................................... 293

Table 92: SIP Settings/General ................................................................................................................. 294

Table 93: SIP Settings/Misc ...................................................................................................................... 294

Table 94: SIP Settings/Session Timer ....................................................................................................... 295

Table 95: SIP Settings/TCP and TLS ........................................................................................................ 296

Table 96: SIP Settings/NAT ....................................................................................................................... 297

Table 97: SIP Settings/ToS ........................................................................................................................ 297

Table 98: PBX Interface Settings .............................................................................................................. 302

Table 99: PMS Supported Features .......................................................................................................... 307

Table 100: PMS Basic Settings ................................................................................................................. 308

Table 101: PMS Wake up Service ............................................................................................................. 310

Table 102: Create New Mini Bar ............................................................................................................... 311

Table 103: Create New Maid ..................................................................................................................... 312

Table 104: SugarCRM Settings ................................................................................................................. 315

Table 105: Salesforce Settings.................................................................................................................. 316

Table 106: Wakeup Service ...................................................................................................................... 319

Table 107: Announcements Center Settings ............................................................................................. 320

Table 108: Group Settings......................................................................................................................... 321

Table 109: Trunk Status ............................................................................................................................ 324

Table 110: Extension Status ...................................................................................................................... 325

Table 111: Interface Status Indicators ....................................................................................................... 326

Table 112: System StatusGeneral ......................................................................................................... 327

Table 113: System StatusNetwork ......................................................................................................... 328

Table 114: CDR Filter Criteria ................................................................................................................... 334

Table 115: CDR Statistics Filter Criteria .................................................................................................... 341

Table 116: API Configuration Files ............................................................................................................ 342

Table 117: User ManagementCreate New User .................................................................................... 348

Table 118: Operation Log Column Header ................................................................................................ 351

Table 119: Change Binding Email option .................................................................................................. 353

Table 120: Network Upgrade Configuration .............................................................................................. 356

Table 121: Data Sync Configuration ......................................................................................................... 362

Table 122: Cleaner Configuration ............................................................................................................. 365

Table 123: USB/SD Card Files Cleanup ................................................................................................... 367

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Table 124: Ethernet Capture ..................................................................................................................... 368

Table of Figures

Figure 1: UCM6102 Front View ................................................................................................................... 37

Figure 2: UCM6102 Back View ................................................................................................................... 38

Figure 3: UCM6104 Front View ................................................................................................................... 38

Figure 4: UCM6104 Back View ................................................................................................................... 39

Figure 5: UCM6108 Front View ................................................................................................................... 40

Figure 6: UCM6108 Back View ................................................................................................................... 40

Figure 7: UCM6116 Front View ................................................................................................................... 40

Figure 8: UCM6116 Back View ................................................................................................................... 40

Figure 9: UCM6104 Web GUI Login Page .................................................................................................. 45

Figure 10: UCM6100 Setup Wizard ............................................................................................................ 46

Figure 11: UCM6100 Web GUI Language .................................................................................................. 47

Figure 12: UCM6102 Network Interface Method: Route ............................................................................. 55

Figure 13: UCM6102 Network Interface Method: Switch ............................................................................ 56

Figure 14: UCM6102 Network Interface Method: Dual ............................................................................... 57

Figure 15: DHCP Client List ........................................................................................................................ 57

Figure 16: Add MAC Address Bind ............................................................................................................. 58

Figure 17: Batch Add MAC Address Bind ................................................................................................... 58

Figure 18: UCM6100 Using 802.1X as Client ............................................................................................. 59

Figure 19: UCM6100 Using 802.1X EAP-MD5 ........................................................................................... 59

Figure 20: UCM6104 Static Route Sample ................................................................................................. 61

Figure 21: UCM6104 Static Route Configuration........................................................................................ 62

Figure 22: Create New Port Forwarding ..................................................................................................... 63

Figure 23: UCM6102 Port Forwarding Configuration ................................................................................. 64

Figure 24: GXP2160 Web Access Using UCM6102 Port Forwarding ........................................................ 64

Figure 25: OpenVPN feature on the UCM6100 .......................................................................................... 65

Figure 26: Register Domain Name on noip.com ......................................................................................... 66

Figure 27: UCM6100 DDNS Setting ........................................................................................................... 67

Figure 28: Using Domain Name to Connect to UCM6100 .......................................................................... 67

Figure 29: Create New Firewall Rule .......................................................................................................... 70

Figure 30: Configure Dynamic Defense ...................................................................................................... 72

Figure 31: Fail2ban Settings ....................................................................................................................... 73

Figure 32: SSH Access ............................................................................................................................... 75

Figure 33: LDAP Server Configurations ...................................................................................................... 76

Figure 34: Default LDAP Phonebook DN .................................................................................................... 76

Figure 35: Default LDAP Phonebook Attributes .......................................................................................... 77

Figure 36: LDAP ServerLDAP Phonebook .............................................................................................. 77

Figure 37: Add LDAP Phonebook ............................................................................................................... 78

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Figure 38: Edit LDAP Phonebook ............................................................................................................... 78

Figure 39: Import Phonebook...................................................................................................................... 78

Figure 40: Phonebook CSV File Format ..................................................................................................... 79

Figure 41: LDAP Phonebook After Import ................................................................................................... 79

Figure 42: Export Selected LDAP Phonebook ............................................................................................ 80

Figure 43: LDAP Client Configurations ....................................................................................................... 81

Figure 44: GXP21XX LDAP Phonebook Configuration .............................................................................. 82

Figure 45: UCM6100 Email Settings ........................................................................................................... 84

Figure 46: Email Templates ......................................................................................................................... 85

Figure 47: Conference Schedule Template ................................................................................................. 86

Figure 48: Email Send log ........................................................................................................................... 87

Figure 49: Email Logs ................................................................................................................................. 88

Figure 50: Set Time Manually ..................................................................................................................... 89

Figure 51: Create New Office Time ............................................................................................................. 90

Figure 52: System SettingsTime SettingsOffice Time.......................................................................... 91

Figure 53: Create New Holiday ................................................................................................................... 92

Figure 54: System SettingsTime SettingsHoliday ................................................................................ 93

Figure 55: PBX SettingsRecordings Storage .......................................................................................... 94

Figure 56: Recordings Storage Prompt Information ................................................................................... 94

Figure 57: Recording Storage Category ..................................................................................................... 95

Figure 58: Google Service SettingsOAuth2.0 Authentication .................................................................. 96

Figure 59: Google ServiceNew Project ................................................................................................... 96

Figure 60: Google ServiceCreate New Credential .................................................................................. 97

Figure 61: Google ServiceOAuth2.0 Login .............................................................................................. 97

Figure 62: Zero Config Configuration Architecture for End Point Device ................................................... 99

Figure 63: UCM6100 Zero Config ............................................................................................................. 100

Figure 64: Auto Provision Settings ............................................................................................................ 101

Figure 65: Auto Discover ........................................................................................................................... 103

Figure 66: Discovered Devices ................................................................................................................. 103

Figure 67: Device list - CSV file sample .................................................................................................... 104

Figure 68: Managing Discovered Devices ................................................................................................ 104

Figure 69: Global Policy Categories ......................................................................................................... 106

Figure 70: Edit Global Template ................................................................................................................ 114

Figure 71: Edit Model Template ................................................................................................................ 116

Figure 72: Template Management ............................................................................................................ 117

Figure 73: Upload Model Template Manually ............................................................................................ 117

Figure 74: Create New Device .................................................................................................................. 119

Figure 75: Manage Devices ...................................................................................................................... 119

Figure 76: Edit Device ............................................................................................................................... 120

Figure 77: Edit Customize Device Settings ............................................................................................... 122

Figure 78: Add P Value in Customize Device Settings ............................................................................. 123

Figure 79: Modify Selected Devices - Same Model .................................................................................. 124

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Figure 80: Modify Selected Devices - Different Models ............................................................................ 125

Figure 81: Device List in Zero Config ........................................................................................................ 126

Figure 82: Zero Config Sample - Global Policy......................................................................................... 127

Figure 83: Zero Config Sample - Device Preview 1 .................................................................................. 128

Figure 84: Zero Config Sample - Device Preview 2 .................................................................................. 129

Figure 85: Zero Config Sample - Device Preview 3 .................................................................................. 130

Figure 86: Create New Device .................................................................................................................. 131

Figure 87: Manage Extensions ................................................................................................................. 155

Figure 88: Export Extensions .................................................................................................................... 156

Figure 89: Import Extensions .................................................................................................................... 157

Figure 90: Import File ................................................................................................................................ 157

Figure 91: Import Error .............................................................................................................................. 163

Figure 92: E-mail Notification - Prompt Information .................................................................................. 164

Figure 93: Account Registration Information and QR Code ...................................................................... 164

Figure 94: LDAP Client Information and QR Code ................................................................................... 165

Figure 95: Multiple Registrations per Extension ....................................................................................... 165

Figure 96: Extension - Concurrent Registration ........................................................................................ 166

Figure 97: SMS Message Support ............................................................................................................ 166

Figure 98: Edit Extension Group ............................................................................................................... 167

Figure 99: Select Extension Group in Outbound Route ............................................................................ 168

Figure 100: UCM6100 FXO Tone Settings ............................................................................................... 172

Figure 101: UCM6100 PSTN Detection .................................................................................................... 172

Figure 102: UCM6100 PSTN Detection: Auto Detect ............................................................................... 173

Figure 103: UCM6100 PSTN Detection: Semi-Auto Detect ..................................................................... 173

Figure 104: DOD extension selection ....................................................................................................... 186

Figure 105: Edit DOD ................................................................................................................................ 186

Figure 106: SLA Station ............................................................................................................................ 187

Figure 107: Enable SLA Mode for Analog Trunk ....................................................................................... 188

Figure 108: Analog Trunk with SLA Mode Enabled .................................................................................. 188

Figure 109: SLA Example - SLA Station ................................................................................................... 188

Figure 110: SLA Example - MPK Configuration ........................................................................................ 189

Figure 111: Outbound Blacklist ................................................................................................................. 193

Figure 112: Create New PIN Group .......................................................................................................... 194

Figure 113: PIN members ......................................................................................................................... 195

Figure 114: Outbound PIN ........................................................................................................................ 195

Figure 115: CDR Record ........................................................................................................................... 195

Figure 116: Importing PIN Groups from CSV files .................................................................................... 196

Figure 117: Incorrect CSV File .................................................................................................................. 196

Figure 118: CSV File Format ..................................................................................................................... 197

Figure 119: CSV File Successful Upload .................................................................................................. 197

Figure 120: Inbound Route feature: Prepend ........................................................................................... 200

Figure 121: Inbound Route - Multiple Mode.............................................................................................. 201

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Figure 122: Inbound Route - Multiple Mode Feature Codes ..................................................................... 202

Figure 123: Blacklist Configuration Parameters........................................................................................ 203

Figure 124: Blacklist csv File..................................................................................................................... 203

Figure 125: Conference ............................................................................................................................ 207

Figure 126: Conference Invitation From Web GUI .................................................................................... 208

Figure 127: Conference Recording ........................................................................................................... 210

Figure 128: Conference Schedule ............................................................................................................ 214

Figure 129: Create New IVR ..................................................................................................................... 215

Figure 130: Key Pressing Events .............................................................................................................. 218

Figure 131: Black/White List ..................................................................................................................... 219

Figure 132: Click on Prompt to Create IVR Prompt .................................................................................. 220

Figure 133: Language Settings for Voice Prompt ..................................................................................... 221

Figure 134: Voice Prompt Package List .................................................................................................... 222

Figure 135: New Voice Prompt Language Added ..................................................................................... 222

Figure 136: Upload Single Voice Prompt for Entire Language Pack ........................................................ 223

Figure 137: Voicemail Settings.................................................................................................................. 224

Figure 138: Voicemail Count ..................................................................................................................... 227

Figure 139: Voicemail Email Settings ....................................................................................................... 228

Figure 140: Voicemail Group..................................................................................................................... 229

Figure 141: Ring Group ............................................................................................................................. 230

Figure 142: Ring Group Configuration ...................................................................................................... 232

Figure 143: Sync LDAP Server option ...................................................................................................... 233

Figure 144: Manually Sync LDAP Server ................................................................................................. 234

Figure 145: Ring Group Remote Extension .............................................................................................. 234

Figure 146: Paging/Intercom Group .......................................................................................................... 235

Figure 147: Page/Intercom Group Settings .............................................................................................. 236

Figure 148: Call Queue ............................................................................................................................. 237

Figure 149: Static Agents limitation ........................................................................................................... 240

Figure 150: Agent Login Settings .............................................................................................................. 241

Figure 151: Call Queue Statistics ............................................................................................................. 243

Figure 152: Call Queue Switchboard ........................................................................................................ 244

Figure 153: Edit Pickup Group .................................................................................................................. 246

Figure 154: Edit Pickup Feature Code ...................................................................................................... 247

Figure 155: Music On Hold Default Class ................................................................................................. 248

Figure 156: Play Custom Prompt .............................................................................................................. 249

Figure 157: Information Prompt ................................................................................................................ 249

Figure 158: Record Custom Prompt ......................................................................................................... 250

Figure 159: Fax Settings ........................................................................................................................... 251

Figure 160: Configure Analog Trunk without Fax Detection ..................................................................... 253

Figure 161: Configure Extension for Fax Machine: FXS Extension ......................................................... 254

Figure 162: Configure Extension for Fax Machine: Analog Settings ........................................................ 254

Figure 163: Configure Inbound Rule for Fax ............................................................................................. 255

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Figure 164: Create Fax Extension ............................................................................................................ 255

Figure 165: Inbound Route to Fax Extension ........................................................................................... 256

Figure 166: Fax Sending in Web GUI ....................................................................................................... 257

Figure 167: Fax Send Progress ................................................................................................................ 257

Figure 168: SIP Presence Configuration .................................................................................................. 259

Figure 169: SIP Presence Feature Code .................................................................................................. 260

Figure 170: Edit Follow Me ....................................................................................................................... 261

Figure 171: Speed Dial Destinations ........................................................................................................ 263

Figure 172: Create New DISA................................................................................................................... 264

Figure 173: Create New Event List ........................................................................................................... 268

Figure 174: Create Dial by Name Group .................................................................................................. 270

Figure 175: Configure Extension First Name and Last Name .................................................................. 271

Figure 176: Dial By Name Group In IVR Key Pressing Events ................................................................ 272

Figure 177: Dial By Name Group In Inbound Rule ................................................................................... 272

Figure 178: StatusPBX StatusActive Calls - Ringing ......................................................................... 274

Figure 179: StatusPBX StatusActive Calls – Call Established .......................................................... 274

Figure 180: call connection less than half hour......................................................................................... 275

Figure 181: call connection between half an hour and one hour .............................................................. 275

Figure 182: Call Connection More Than One Hour .................................................................................. 276

Figure 183: Configure to Monitor an Active Call ....................................................................................... 276

Figure 184: Enable/Disable Feature codes ............................................................................................... 283

Figure 185: Download Recording File from CDR Page ............................................................................ 284

Figure 186: Download Recording File from Recording Files Page ........................................................... 284

Figure 187: Record New Custom Prompt ................................................................................................. 288

Figure 188: Upload Custom Prompt ......................................................................................................... 289

Figure 189: Download All Custom Prompt ................................................................................................ 289

Figure 190: SIP Trunk Prompt Tone .......................................................................................................... 299

Figure 191: Transparent Call-Info ............................................................................................................. 300

Figure 192: FXS Ports Signaling Preference ............................................................................................ 301

Figure 193: FXO Ports ACIM Settings ...................................................................................................... 301

Figure 194: DAHDI Settings ...................................................................................................................... 303

Figure 195: CTI Server Listening port ....................................................................................................... 304

Figure 196: UCM & PMS interaction ......................................................................................................... 306

Figure 197: UCM & PMS interaction ......................................................................................................... 307

Figure 198: Create New Room ................................................................................................................. 309

Figure 199: Room Status .......................................................................................................................... 309

Figure 200: Add batch rooms .................................................................................................................... 310

Figure 201: Create New Wake Up Service ............................................................................................... 310

Figure 202: Wakeup Call executed ........................................................................................................... 311

Figure 203: Create New Mini Bar .............................................................................................................. 311

Figure 204: Create New Maid ................................................................................................................... 312

Figure 205: Create New Consumer Goods ............................................................................................... 312

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Figure 206: Mini Bar .................................................................................................................................. 313

Figure 207: SugarCRM Basic Settings ..................................................................................................... 314

Figure 208: CRM User Settings ................................................................................................................ 315

Figure 209: Salesforce Basic Settings ...................................................................................................... 316

Figure 210: Salesforce User Settings ....................................................................................................... 317

Figure 211: Create New Wakeup Service - Admin Login .......................................................................... 318

Figure 212: Wakeup Service Feature Code .............................................................................................. 319

Figure 213: Announcements Center ......................................................................................................... 320

Figure 214: Announcements Center Group Configuration ........................................................................ 321

Figure 215: Announcements Center Code Configuration ......................................................................... 322

Figure 216: Announcements Center Example .......................................................................................... 322

Figure 217: StatusPBX Status ............................................................................................................... 323

Figure 218: Trunk Status ........................................................................................................................... 324

Figure 219: Extension Status .................................................................................................................... 325

Figure 220: UCM6102 Interfaces Status ................................................................................................... 326

Figure 221: System StatusStorage Usage ............................................................................................ 329

Figure 222: System StatusResource Usage ......................................................................................... 329

Figure 223: System EventsAlert Events Lists: Disk Usage ................................................................... 330

Figure 224: System EventsAlert Events Lists: External Disk Usage..................................................... 331

Figure 225: System EventsAlert Events Lists: Memory Usage ............................................................. 331

Figure 226: System EventsAlert Events Lists: System Crash ............................................................... 332

Figure 227: System EventsAlert Log ..................................................................................................... 332

Figure 228: Filter for Alert Log .................................................................................................................. 333

Figure 229: CDR Filter .............................................................................................................................. 334

Figure 230: Call Report ............................................................................................................................. 336

Figure 231: Call Report Entry with Audio Recording File .......................................................................... 337

Figure 232: Automatic Download Settings ................................................................................................ 338

Figure 233: CDR Report ........................................................................................................................... 338

Figure 234: Detailed CDR Information ...................................................................................................... 338

Figure 235: Downloaded CDR File Sample .............................................................................................. 339

Figure 236: Downloaded CDR File Sample - Source Channel and Dest Channel 1 ................................ 339

Figure 237: Downloaded CDR File Sample - Source Channel and Dest Channel 2 ................................ 340

Figure 238: CDR Statistics ........................................................................................................................ 341

Figure 239: CDRRecording Files........................................................................................................... 342

Figure 240: Edit User Information by Super Admin .................................................................................. 344

Figure 241: User Portal Login ................................................................................................................... 345

Figure 242: User Portal Layout ................................................................................................................. 345

Figure 243: User Management Page Display ........................................................................................... 347

Figure 244: Create New User ................................................................................................................... 347

Figure 245: User Management – New Users ............................................................................................ 348

Figure 246: General User .......................................................................................................................... 349

Figure 247: Create New Custom Privilege ................................................................................................ 350

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Figure 248: Multiple User Operation Error Prompt ................................................................................... 350

Figure 249: Operation Logs ...................................................................................................................... 351

Figure 250: Operation Logs Filter ............................................................................................................. 352

Figure 251: Change Password.................................................................................................................. 353

Figure 252: Change Binding Email ........................................................................................................... 353

Figure 253: Login Timeout Settings .......................................................................................................... 355

Figure 254: Network Upgrade ................................................................................................................... 356

Figure 255: Local Upgrade........................................................................................................................ 357

Figure 256: Upgrading Firmware Files ...................................................................................................... 357

Figure 257: Reboot UCM6100 .................................................................................................................. 358

Figure 258: Create New Backup ............................................................................................................... 359

Figure 259: Restore Warning .................................................................................................................... 360

Figure 260: Backup / Restore ................................................................................................................... 360

Figure 261: Local Backup ......................................................................................................................... 361

Figure 262: Data Sync .............................................................................................................................. 362

Figure 263: Restore UCM6100 from Backup File ..................................................................................... 363

Figure 264: Reset and Reboot .................................................................................................................. 364

Figure 265: Cleaner .................................................................................................................................. 365

Figure 266: USB/SD Card Files Cleanup .................................................................................................. 366

Figure 267: Ethernet Capture.................................................................................................................... 368

Figure 268: PING ...................................................................................................................................... 369

Figure 269: Traceroute .............................................................................................................................. 369

Figure 270: Troubleshooting Analog Trunks ............................................................................................. 370

Figure 271: A Key Dial-up FXO ................................................................................................................. 371

Figure 272: Service Check ........................................................................................................................ 372

Figure 273: Network Status....................................................................................................................... 372

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DOCUMENT PURPOSE

This document describes UCM6100 series specifications, features and will help you to configure your system via Web GUI menu to fully manipulate the supported features. The intended audiences of this document are device administrators. To learn more about UCM6100 series features, please visit http://www.grandstream.com/support to download available how-to guides.

This guide covers following topics:

Product overview

Installation

Getting started

System settings

Provisioning

Extensions

Extension groups

Analog Trunks

VoIP Trunks

SLA station

Call routes

Conference bridge

Conference schedule

IVR

Language settings for voice prompt

Voicemail

Ring group

Paging and intercom group

Call queue

Pickup groups

PIN Groups

Music on hold

Fax Server

Busy camp-on

Presence

Follow me

Speed Dial

DISA

Callback

BLF and event list

Dial by name

Active calls and monitor

Call features

Call recording

CTI Server

Asterisk manager interface (AMI)

CRM integration

PMS integration

Wakeup service

Announcements center

Status and reporting

CDR (Call Details Record)

User Portal

Upgrading and maintenance

Backup/restore

Troubleshooting

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CHANGE LOG

This section documents significant changes from previous versions of the UCM6100 user manuals. Only major new features or major document updates are listed here. Minor updates for corrections or editing are not documented here.

Firmware Version 1.0.15.16

• Added support to announce name in Dial By Name feature. [DIAL BY NAME]

• Increased maximum number of call queue static agents. [Static Agents limitation]

• Added Queue Log Cleaner. [Queue Log Cleaner]

• Added operation logs details and remarks. [Operation Log]

• Added extension level voicemail-to-Email setting. [Send Voicemail to Email] [Keep Voicemail after

Emailing]

• Added support for reset certificate. [Reset Certificates]

• Added support for GXP21xx color phone queue login/logout softkey. [Enable Agent Login]

• Added support to add comments to inbound/outbound route patterns. [Pattern] [Pattern]

• Added PPI mode option under SIP trunk advanced settings [PPI Mode]

• Added ability to import PIN groups from CSV files. [Importing PIN Groups from CSV files]

• Added support for wakeup groups. [WAKEUP SERVICE]

• Added support for SYN Flood defense. [SYN-Flood Defense Enable]

• Added Queue Log option to Backup/Restore page. [Backup/Restore]

• Modified Switchboard UI to offer easier access to call Options. [Switchboard]

• Further optimized Call Queue Statistics page to provide a more user-friendly experience and improve

performance. [Queue Statistics]

• Added image uploading support to Email templates [Email Templates]

• Added IPv6 gateway support. [Static Routes]

• Improved Voice Message Responses for failed SIP trunk calls. [SIP Trunk Prompt Tone]

• Added Fail2Ban to support TCP/TLS beside the already-supported UDP. [Fail2ban]

• Restored ability to sort the ringing order of Follow Me numbers. [FOLLOW ME]

• Added option to restrict users from changing their SIP credentials through user portal [Consumer

Privilege]

• Added option to enable or disable fax-to-mail feature on extension level. [Fax To Email]

• Added IP address whitelist to web GUI configuration. [Enable IP whitelist]

Firmware Version 1.0.14.24

• Added protection to prevent HTTP rogue login.

Firmware Version 1.0.14.23

• Restored ability to view voicemail count in the Extension/Trunk overview. [Extension Voicemail Count]

• Restored the ability to set custom numbers for call forwarding settings. [Call Forward Unconditional]

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[Call Forward No Answer] [Call Forward Busy]

• Restored previous format for entering multiple dial plans (one pattern per line) for inbound/outbound

rules. [Pattern] [Pattern]

• Restored Zero Config’s sorting by column and introduced a search bar. [Managing discovered devices]

Firmware Version 1.0.14.21

• A new web GUI with consistent operating style, can display/update the system's status in real time.

• Added support for SIP Presence. [PRESENCE]

• Added support for CallCenter feature/ Virtual Call Queue. [Call Center Settings & Enhancement]

• Added support for Call Queue position announcement. [Call Center Settings & Enhancement]

• Added support for Call Queue Statistics. [Queue Statistics]

• Added support for Call Queue Auto-Fill. [Queue Auto fill enhancement]

• Added Switchboard for call queue monitoring. [Switchboard]

• Added ability to restore blind transfer call to transferrer. [Allow callback when blind transfer fails]

• Added support for external disk cleaner. [Cleaner]

• Added option to enable DOD when call is being diverted/forwarded. [Use callee DOD on FWD or Ring

Simultaneously]

• Change follow me settings to extension level settings. [FOLLOW ME]

• Added support for call forward whitelist. [FWD Whitelist]

• Added Fail2Ban defense from web login attack. [Login Attack Defense]

• Added limitation for maximum number of call queue static agents. [Static Agents limitation]

• Added Support for wakeup service module in Custom privilege. [Custom Privilege]

• Added IPv6 support for T.38.

• Added DAHDI settings. [DAHDI Settings]

• Added ability to pass through SIP Call-Info header to support GXP phone JPEG_Over_HTTP with

encryption and authentication to open door for GDS3710. [Transparent Call-Info header]

Firmware Version 1.0.13.14

• Added extension whitelist/blacklist for IVR dialing. [IVR]

• Added ability to include DOD in PPI Header for SIP trunk. [PPI Mode]

• Added ability to customize PAI Header. [PAI Header]

• Added blacklist for outbound calls. [Outbound Blacklist]

• Added support to upload/download MOH package from Web GUI. [MUSIC ON HOLD]

• Added support to download custom prompts from Web GUI. [Download All Custom Prompt]

Added option to configure prompt timeout in Dial By Name. [DIAL BY NAME]

• Added description field in ZeroConfig settings to configure Softkey/Line/MPK for GXP series phones.

[PROVISIONING]

• Improved seamless transfer privilege control. [Seamless transfer privilege control]

• Added RTP Keep-alive support. [RTP Keep-alive]

• Added Email Send Log. [Email Send Log]

• Added support for Mitel simulation/protocol interfaces for PMS module. [PMS]

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• Added support for up to 10 failover trunks. [Use Failover Trunk]

Firmware Version 1.0.12.19

• Added support for binding a mobile phone number to extension [Mobile Phone Number]

• Added support opus codec

• Added support call-barging privilege settings based on extensions [Monitor privilege control]

• Added support for Seamless Transfer [Enable Seamless Transfer]

• Added support for Custom Call-Info for Auto Answer [Custom Call-info for Auto Answer]

• Added support for DND Whitelist [DND Whitelist]

• Add the Field Description on Softkey, Line keys and MPK from Zero Config

• Added support to select interval for numbers on Batch add extension [Extension ]

• Added support for Batch Add CallerID Number [CallerID Number]

• Added support for Search Extensions Using CallerID Name

• Added support to Enable/Disable Inbound and Outbound Route [Disable This Route] [Disable This

Route]

• Added support for Outbound Route Time Condition [Time Condition]

• Added support for IPv6 [IPv6 Address]

• Added Support for MTU configurable [MTU]

• Added support of CRM [CRM]

• Added support for Custom Privilege in User Management [Custom Privilege]

• Added Hotline support for FXS Extension [Hotline]

• Added support for Separate Wakeup Service [WAKEUP SERVICE]

• Added ability to provision phones from different network subnets using zero config [Subnet Whitelist]

• One-key-dial is replaced by Speed Dial to support more than one digit [SPEED DIAL]

• Added Append extension number in the end of DOD [Direct Outward Dialing (DOD)]

• Support Japan CID NTT Detect

• Added support for Ethernet Capture Auto Sync to SFTP Server [Enable SFTP Data Sync]

• Added support for Ethernet Capture saved to External Storage Device [Storage to External Device]

• Added support for Disable Extension Range on the Setup Wizard [Setup Wizard]

• Added more support for Port Forwarding [Port Forwarding (UCM6102 only)]

• Added support for USB/SD Card Files Cleanup [USB/SD Card Files Cleanup]

• Added support for A Key Dial-up FXO [A key Dial-up FXO]

• Added support for ACIM Detect Option for FXO [FXO Ports ACIM Settings]

• Added support for some special character on the file name of FW [Upgrading via Local Upload]

• Added more search criteria of CDR [CDR]

• Added support of "Allow outgoing calls if registration failure" for register trunks [Allow outgoing calls if registration failure]

• Added support for music on hold playback from webGUI [MUSIC ON HOLD]

• Added support to enable delete recording files for user privilege [Consumer Privilege]

• Added support disk Inode usage in "Storage Usage" page [Storage Usage]

• Added support foe Ring Group/Call Queue/IVR Display Option for Caller ID [Replace Display Name |

Replace Caller ID | Replace Caller ID]

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• Added support for compatibility between backup package from UCM61xx and UCM62xx

[Backup/Restore]

• Added support to Detect talking users in conference [CONFERENCE ]

• Added Support of Mini Bar for PMS [Mini Bar]

Firmware Version 1.0.11.27

• Added ability to sort extension status on Web GUI [Extensions]

• Added one click enable / disable feature code [Feature Codes]

• Added Uruguay time zone support [Auto Time Updating]

• Added distinctive ring tone support [Configure Call Queue][Configure IVR][Create New SIP Extension]

• Added special character support for SFTP client account [Data Sync]

• Added destination directory support for data sync [Data Sync]

• Added ring group music on hold [Configure Ring Group]

• Added CDR multi-email / time condition support [CDR]

• Added blacklist anonymous call block [Blacklist Configurations]

• Added ability to sort selected extension in Eventlist [Event List]

• Added Banned User list for Web GUI login attempts [Login Settings]

• Added Email template support [Email Templates]

• Added outbound route country restriction

• Added external disk usage alert option [Alert Events List]

• Added range IP input support for dynamic defense white list [Dynamic Defense]

• Added blacklist support for Fail2ban [Fail2ban]

• Added ability to reboot device from zero config page [Discovery]

• Added GXP1628B template for zero config [Model Update]

• Added PIN group support [PIN Groups]

• Added PMS support [PMS]

• Added call queue custom prompt support [Configure Call Queue]

• Added call queue retry time support [Configure Call Queue]

• Added Support for DHCP Client List [DHCP Client List]

Firmware Version 1.0.10.44

• Added Zero Config DP750 support. [Model Templates]

Firmware Version 1.0.10.39

• Added multiple modes support for inbound route. [Inbound Route: Multiple Mode]

• Added option “Enable Inbound Multiple Mode”, “Inbound Default Mode” and “Inbound Mode 1” for

switching inbound route mode via feature code. [Feature Codes]

• Added prepending prefix for inbound route. [Inbound Route: Prepend Example]

• Added multiple registration per extension. [Multiple Registrations per Extension]

• Added SIP Message support. [SMS Message Support]

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• Added 100rel option for 100rel support. [Table 97: SIP Settings/ToS]

• Added video preview support.

• Added User Portal Page Fax sending support.

• Added Fax intelligent routing. [Fax Intelligent Route]

• Added Re-Invite with two media (audio, image) support for fax sending. [Fax with Two Media]

• Added option “Max Concurrent Sending Fax” in Fax settings. [Configure Fax/T.38]

• Added option “Fax Queue Length” in Fax settings. [Configure Fax/T.38]

• Added Google Service Setting Support. [Google Service Settings Support]

• Added Conference Schedule. [CONFERENCE SCHEDULE]

• Added Setup Wizard. [Setup Wizard]

• Added ability to customize specific prompt. [Customize Specific Prompt]

• Added option “ALL” when making backup file. [Backup/Restore]

• Added “Enable Destination” and “Default Destination” in Follow Me settings. [FOLLOW ME]

• Added “Call Duration Limit” option in Web GUIPBXInternal OptionsGeneral. [General]

• Added “Enable Auto E-mail Notification” option in Web GUIPBXInternal OptionsGeneral.

[General]

• Added options “ICE Support” and “STUN Server” in Web GUIPBXInternal OptionsRTP

Settings. [PBX Settings/RTP Settings]

• Added payload type setting for VP8 in Web GUIPBXInternal OptionsPayload. [PBX

Settings/Payload]

• Added options “External Host” and “Use IP address in SDP” in Web GUIPBXSIP SettingsNAT.

[SIP Settings/NAT]

• Improved CDR. [CDR Improvement]

• Added Network Status page under Web GUIMaintenanceTroubleshootingNetwork Status

[Network Status]

Firmware Version 1.0.9.26

• Added Announcements Center. [ANNOUNCEMENTS CENTER]

Firmware Version 1.0.9.25

• Added UCM Zero Config model templates for provisioning GXP2200EXT coupled with GXV3240.

[Model Templates]

• Added ability to select the voicemail storage (Support Email + WAV). [Voicemail Email Settings]

Firmware Version 1.0.9.23

• Added SSH enable/disable option in LCD or WEB [SSH Access]

• Added remote peer extensions in Ring Group [Remote Extension in Ring Group]

• Added ability to prepend digits in inbound routes [Table 61: Inbound Rule Configuration Parameters]

• Added ability to search extensions on Extension page. [Search and Edit Extension]

• Added user portal for users to log in with extension number, access user information, extension configuration and CDR.

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• Added support to send Fax via Web GUI. [Fax Sending]

• Added "Enable LDAP" option to skip the extension from UCM default LDAP phonebook. [Table 33: SIP

Extension Configuration ParametersBasic Settings]

• Added DDNS Support. [DDNS Settings]

• Added support for Call Barging using feature codes. [Enable Spy]

• Added ability to search the CDR by called number. [Table 114: CDR Filter Criteria]

• Added ability to select the file types for automatic backup. [Backup/Restore]

• Added automatic backup support on SD Card or USB storage. [Backup/Restore]

• Added support to skip trunk authentication by time condition. [Table 35: SIP Extension Configuration

ParametersFeatures]

• Added option to send P-Asserted-Identity header in SIP Register Trunk. [Table 53: SIP Register Trunk

Configuration Parameters]

• Added ability to specify trunks in CDR filters. [CDR]

• Added ability to use Pattern in Caller Number to filter CDR. [CDR]

• Added support to send UNREGISTER when VoIP trunk is disabled. [Table 53: SIP Register Trunk

Configuration Parameters]

• Added LDAP Client support [LDAP Client Configurations]

• Added option to specify the chronological order of voice mails. [Table 67: Voicemail Settings]

• Added option to configure whether to skip pressing 1/2 to accept or reject calls from Follow Me. [Table

79: Follow Me Settings]

• Added option to specify port range in Port Forwarding configuration. [Table 11: UCM6102 Network

SettingsPort Forwarding]

• Added ability to go back to IVR menu from Dial By Name by pressing the star key. [Dial By Name

Configuration]

• Added support to upgrade SIP end device via SD card in Zero Config. [Table 28: Global Policy

ParametersMaintenance]

• Added ability to filter alert logs. [Alert Log]

• Added ability to delete alert logs. [Alert Log]

• Added NAT option for peer trunk. [SIP Peer Trunk Configuration Parameters]

• Improved Automatic Download CDR result format. [CDR]

Firmware Version 1.0.8.12

• Added recording files store management. [Recordings Storage]

• Added AMI access (restricted). [ASTERISK MANAGER INTERFACE]

• Added LDAP Phonebook Import/Export support. [LDAP Phonebook]

• Added Callback feature. [CALLBACK FEATURES]

• Added Busy Camp-on (Call Completion) feature. [BUSY CAMP-ON]

• Added external number calling via extension. [Table 35: SIP Extension Configuration

ParametersFeatures]

• Added end point firmware management support to allow end point device to upgrade via UCM6100

local storage or USB media. [Table 28: Global Policy ParametersMaintenance]

• Added link to access phone web GUI in Zero Config. [Manage Devices]

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• Added model update support to download model templates for surveillance and GSWave. [Model

Update]

• Added filter for searching in Alert Log. [Alert Log]

• Added DAHDI Out Line Selection. [Table 50: Analog Trunk Configuration Parameters]

• Added automatic CDR download. [CDR]

• Added multi-user management and concurrent user login support. [User Management]

• Added operation log. [Operation Log]

• Added option to specify web access session timer. [Login Settings]

• Added call monitor/barge-in support.

• Added GSwave QR code support. [E-mail Notification]

• Added ability to add customized Pvalue on Model Template. [Model Templates]

• Updated maximum length of Web GUI login password to 30 characters. [Change Password]

• Added Fax gateway support for Fax mode on FXS and FXO port. [Table 42: FXS Extension

Configuration ParametersMedia] [Table 50: Analog Trunk Configuration Parameters]

• Added “NAT” option and “Send PPI Header” option in SIP register trunk configuration. [Table 53: SIP

Register Trunk Configuration Parameters]

• Added REC API (access for call recording files). [API Configuration]

• Added “Synchronize All Data” option in Data Sync. [Data Sync]

• Added “Detect Selected Sound Files” option for Music On Hold files. [MUSIC ON HOLD]

Firmware Version 1.0.7.11

• Added office time and holiday setting. [Office Time] [Holiday]

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

• Added Alert-info configuration for distinctive ringing on inbound route. [Table 61: Inbound Rule

Configuration Parameters]

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

• Added Follow Me support. [FOLLOW ME]

• Added One-Key Dial support.

• Added FXO Dial Delay option for analog trunk. [Table 50: Analog Trunk Configuration Parameters]

• Added Music On Hold selection per extension. [EXTENSIONS]

• Added Call Forward Time Condition per extension. [EXTENSIONS]

• Added custom prompt support for ring group announcement. [Table 71: Ring Group Parameters]

• Added custom prompt support for paging/intercom group announcement. [Table 72: Paging/Intercom

Group Configuration Parameters]

• Added options to delete voicemail and Fax files via cleaner. [System Clean]

• Added DTMF configuration per trunk. [VOIP TRUNKS]

• Added call queue Music On Hold customization. [Table 73: Call Queue Configuration Parameters]

• Added call queue destination support. [Table 73: Call Queue Configuration Parameters]

• Added batch add blacklist file support. [Blacklist Configurations]

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Firmware Version 1.0.6.11

• Added service check options under Web GUIMaintenanceTroubleshootingService Check page to regularly check Asterisk service.

Firmware Version 1.0.6.10

• Added static routes function. [Static Routes]

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

• Added option to disable extension/trunk.

• Added TEL URL support for extension/trunk.

• Added option to dial trunk password per extensions.

• 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 GUI.

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

• Added payload configuration for audio/video codecs. [PBX Settings/Payload]

• Added activity calls status on Web GUI status page.

• Added CDR API support. [API Configuration]

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

Config, User Login Success, User Login Failed, SIP Internal Call Failure and etc. [Alert Events List]

Firmware Version 1.0.5.19

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

• Added "Direct Dial Voicemail Prefix" feature code back. [Table 84: UCM6100 Feature Codes]

• Changed valid range for option "Current Disconnect Threshold". [Table 50: Analog Trunk Configuration

Parameters]

Firmware Version 1.0.5.14

• New backend data structure and Web GUI 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 GUI. [Web GUI Languages]

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

settings. [Table 59: Outbound Route Configuration Parameters]

• Added more language support for voice prompt. [LANGUAGE SETTINGS]

• Added "Ring Group Destination" for ring group configuration. [Table 71: Ring Group Parameters]

• Added "Extension Groups" section in Web GUI. [EXTENSION GROUPS]

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• Added "Pickup Groups" section in Web GUI. [PICKUP GROUPS]

• Added BLF function description. [BLF AND EVENT LIST]

• Updated default extension range. [Table 85: 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 52: Create New SIP Trunk]

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

• Added GXP one-button Voicemail access. [Table 33: SIP Extension Configuration Parameters]

• Added option "Skip voicemail password verification" on extension edit page. [Table 33: SIP Extension

Configuration Parameters]

• Added Hot-Desking Support. [Table 33: 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 98: ]

• 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

226: 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 52: Create New SIP Trunk]

Firmware Version 1.0.3.13

• Added Fail2Ban support for SIP authentication. [Fail2ban]

• Added "Auto Record" option for trunk.

• Added "Digit Timeout" option and voice prompt "Language" selection for IVR. [Table 66: IVR

Configuration Parameters]

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

[Table 84: UCM6100 Feature Codes]

• Added "Enforce Strong Passwords" option. [Table 85: Internal Options/General]

• Added FXS MWI Mode. [Table 98: PBX Interface Settings]

• 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]

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• 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]

• 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 84: 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 52: Create New SIP Trunk]

• Added LDAP sync options for peer SIP trunk. [Table 52: Create New SIP Trunk]

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

• Added DISA and Fax to IVR key press event options. [Table 66: IVR Configuration Parameters]

• Added "Min Message Time" option in voicemail settings. [Table 67: Voicemail Settings]

• Added Fax setting samples. [FAX SERVER]

• 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 84: UCM6100 Feature Codes]

• Added feature code *8 for "Pickup Extension" in pickup group feature. [Table 84: UCM6100 Feature

Codes]

• Added "Record Prompt" and "Custom Name of Pickup Group" options in internal options. [Table 85:

Internal Options/General]

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

misconfiguration. [Table 89: IAX Settings/General]

• Changed reset mode to two mode "User Data" and "All".

Firmware Version 1.0.1.22

• This is the initial version.

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WELCOME

Thank you for purchasing Grandstream UCM6100 series IP PBX appliance. The UCM6100 series IP PBX appliance is designed to bring enterprise-grade voice, video, data, and mobility features to small-to-medium businesses (SMBs) in an easy-to-manage fashion. This IP PBX series allows businesses to unify multiple communication technologies, such comprehensive voice, video calling, video conferencing, video surveillance, data tools and facility access management onto one common network that that can be managed and/or accessed remotely. The UCM6100 series sports a 1GHz ARM Cortex A8 processor,

512MB RAM and 4GB flash. The secure and reliable UCM6100 series delivers enterprise-grade features without any licensing fees, costs-per-feature or recurring fees.

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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.

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This document is subject to change without notice. The latest electronic version of this user manual is available for download here: http://www.grandstream.com/support

Reproduction or transmittal of the entire or any part, in any form or by any means, electronic or print, for any purpose without the express written permission of Grandstream Networks, Inc. is not permitted.

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PRODUCT OVERVIEW

Technical Specifications

The following table resumes all the technical specifications including the protocols / standards supported, voice codecs, telephony features, languages and upgrade/provisioning settings for UCM6100 series.

Table 1: Technical Specifications

Interfaces

Analog Telephone FXS Ports 2 ports (both with lifeline 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 Gigabit RJ45 ports with integrated PoE Plus

(IEEE 802.3at-2009)

• UCM6108/6116: Single Gigabit RJ45 port with integrated PoE Plus

(IEEE 802.3at-2009)

NAT Router

Peripheral Ports

LED Indicators

Yes, UCM6102 only

USB, SD

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

LCD Display

Reset Switch

128x32 graphic LCD with DOWN and OK button

Yes

Voice/Video Capabilities

Voice-over-Packet

Capabilities

Voice and Fax Codecs

Video Codecs

QoS

Signaling and Control

DTMF 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

(30ms only), GSM, AAL2-G.726-32, ADPCM; T.38

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

Layer 3 QoS, Layer 2 QoS

Provisioning Protocol and

Plug-and-Play

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), eventlist between local and remote trunk

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Network Protocols

Disconnect Methods

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

Dimensions

Environmental

Mounting

Weight

• Output: 12VDC, 1.5A

• Input: 100-240VAC, 50-60Hz

• UCM6102/6104: 226mm (L) x 155mm (W) x 34.5mm (H)

• UCM6108/6116: 440mm (L) x 185mm (W) x 44mm (H)

• Operating: 32 - 104 o F / 0 - 40 o C, 10-90% (non-condensing)

• Storage: 14 - 140 o F / -10 - 60 o C

• UCM6102/6104: Wall mount and Desktop

• UCM6108/6116: Rack mount and Desktop

• 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

Additional Features

Multi-language Support

Caller ID

Polarity Reversal/ Wink

English/Simplified Chinese/Traditional Chinese/Spanish/French/

Portuguese/German/Russian/Italian/Polish/Czech for Web GUI;

Customizable IVR/voice prompts for English, Chinese, British English,

German, Spanish, Greek, French, Italian, Dutch, Polish, Portuguese,

Russian, Swedish, Turkish, Hebrew, Arabic;

Customizable language pack to support any other languages

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

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

Call Center

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

Customizable Auto Attendant Up to 5 layers of IVR (Interactive Voice Response)

Maximum Call Capacity

Conference Bridges

• UCM6102: Up to 30 simultaneous calls

• UCM6104: Up to 45 simultaneous calls

• UCM6108/6116: Up to 60 simultaneous calls

Or up to 66% performance if calls are SRTP encrypted

• UCM6102/6104: Up to 3 password-protected conference bridges allowing up to 25 simultaneous PSTN or IP participants

• UCM6108/6116: Up to 6 password-protected conference bridges allowing up to 32 simultaneous PSTN or IP participants

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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 and ITU-T K.21 (Basic Level)

• UL 60950 (power adapter)

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Note:

• UCM6100 FXS ports lifeline functionality:

• The UCM6100 FXS interfaces are metallic through to the FXO interfaces. If there is power outage,

FXS1 port will fail over to FXO 1 port, FXS 2 port will fail over to FXO 2 port. The user can still access the PSTN connected with the FXO interfaces from FXS interfaces.

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INSTALLATION

Before deploying and configuring the UCM6100 series, the device needs to be properly powered up and connected to network. This section describes detailed information on installation, connection and warranty policy of the UCM6100 series.

Equipment Packaging

Connect Your UCM6100

Connect The UCM6102

Table 2: UCM6102/UCM6104 Equipment Packaging

Main Case

Power Adaptor

Ethernet Cable

Quick Installation Guide

GPL License

Yes (1)

Yes (1)

Yes (1)

Yes (1)

Yes (1)

Figure 1: UCM6102 Front View

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Figure 2: UCM6102 Back View

To set up the UCM6102, follow the steps below:

1.

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

2.

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

3.

Connect the 12V DC power adapter into the 12V DC power jack on the back of the UCM6102. Insert the main plug of the power adapter into a surge-protected power outlet.

4.

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

5.

Once the UCM6102 is successfully connected to network, the LED indicator for WAN in the front will be in solid green and the LCD shows up the IP address.

6.

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

Fax) to the FXS ports.

Connect The UCM6104

Figure 3: UCM6104 Front View

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Figure 4: UCM6104 Back View

To set up the UCM6104, follow the steps below:

1.

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

2.

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

3.

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

4.

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

5.

Once the UCM6104 is successfully connected to network, the LED indicator for LAN 1 in the front will be in solid green and the LCD shows up the IP address.

6.

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

Fax) to the FXS ports.

Connect The UCM6108

To set up the UCM6108, follow the steps below:

1.

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

2.

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

3.

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

4.

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

5.

Once the UCM6108 is successfully connected to network, the LED indicator for NETWORK in the front will be in solid green and the LCD shows up the IP address.

6.

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

Fax) to the FXS ports.

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Figure 5: UCM6108 Front View

Connect The UCM6116

Figure 6: UCM6108 Back View

Figure 7: UCM6116 Front View

Figure 8: UCM6116 Back View

To set up the UCM6116, follow the steps below:

1.

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

2.

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

3.

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

4.

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

5.

Once the UCM6116 is successfully connected to network, the LED indicator for NETWORK in the front will be in solid green and the LCD shows up the IP address.

6.

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

Fax) to the FXS ports.

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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.

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Warning:

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

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GETTING STARTED

The UCM6100 series provides LCD interface, LED indication and web GUI configuration interface.

• The LCD displays hardware, software and network information. Users could also navigate in the LCD menu for device information and basic network configuration.

• The LED indication at the front of the device provides interface connection and activity status.

• The web GUI gives users access to all the configurations and options for UCM6100 series setup.

This section provides step-by-step instructions on how to use the LCD menu, LED indicators and Web GUI of the UCM6100 series. Once the basic settings are done, users could start making calls from UCM6100 extension registered on a SIP phone as described at the end of this section.

Use The LCD Menu

Default LCD Display

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 3: LCD Menu

Options].

Menu Navigation

Press the "Down" arrow key to browser different menu options. Press the "OK" button to select an entry.

Exit

If "Back" option is available in the menu, select it to go back to the previous menu. For "Device Info"

"Network Info" and "Web Info" which do not have "Back" option, simply press the "OK" button to go back to the previous menu. Also, the LCD will display default idle screen after staying in menu option for 15 seconds.

LCD Backlight

The LCD backlight will be on upon key pressing. The backlight will go off after the LCD stays in idle for

30 seconds.

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View Events

Device Info

Network Info

Network Menu

Factory Menu

Table 3: LCD Menu Options

Critical Events

Other Events

Hardware: Hardware version number

Software: Software version number

P/N: Part number

WAN MAC: WAN side MAC address (UCM6102 only)

LAN MAC: LAN side MAC address

Uptime: System up time

For UCM6104/UCM6108/UCM6116:

LAN Mode: DHCP, Static IP, or PPPoE

LAN IP: IP address

LAN Subnet Mask

For UCM6102:

WAN Mode: DHCP, Static IP, or PPPoE

WAN IP: IP address

WAN Subnet Mask

LAN IP: IP address

LAN Subnet Mask

For UCM6104/UCM6108/UCM6116:

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

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 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.

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Web Info

SSH Switch

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 StatusGeneral 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

Enable SSH: Enable SSH access.

Disable SSH: Disable SSH access.

By default, SSH access is disabled.

Use The LED Indicators

The UCM6100 has LED indicators in the front to display connection status. The following table shows the status definitions.

Table 4: UCM6102/UCM6104 LED Indicators

LED Status

LED Indicator

LAN

WAN

USB

SD

FXS (Phone/Fax)

FXO (Telco Line)

Solid: Connected

Flashing: Data Transferring

OFF: Not Connected

LED

NETWORK

Table 5: 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

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Use The Web GUI

Access Web GUI

The UCM6100 embedded Web server responds to HTTP/HTTPS GET/POST requests. Embedded HTML pages allow users to configure the device through a Web browser such as Microsoft IE, Mozilla Firefox,

Google Chrome.

Figure 9: UCM6104 Web GUI Login Page

To access the Web GUI:

1.

Connect the computer to the same network as the UCM6100.

2.

Ensure the device is properly powered up and shows its IP address on the LCD.

3.

Open a Web browser on the computer and enter the web GUI URL in the following format:

http(s)://IP-Address:Port where the IP-Address is the IP address displayed on the UCM6100 LCD.

By default, the protocol is HTTPS and the Port number is 8089.For example, if the LCD shows

192.168.40.167, please enter the following in your web browser:

https://192.168.40.167:8089

4.

Enter the administrator’s login and password to access the Web Configuration Menu. The default administrator's username and password is "admin" and "admin". It is highly recommended to change the default password after login for the first time.

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Note:

By default, the UCM6100 has "Redirect From Port 80" enabled. Therefore, if users type in the UCM6100

IP address in the web browser, the web page will be automatically redirected to the page using HTTPS and port 8089. For example, if the LCD shows 192.168.40.167, please enter 192.168.40.167 in your web browser and the web page will be redirected to: https://192.168.40.167:8089

The option "Redirect From Port 80" can be configured under the UCM6100 web GUISystem

SettingsHTTP Server.

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Setup Wizard

When the user logs in the UCM6100 Web GUI for the first time, a setup wizard will guide the user to set up basic configuration. Configurations in setup wizard includes: Time zone, Change password, Network settings,

Extensions, Trunk and routes.

Figure 10: UCM6100 Setup Wizard

Users can disable the Extension Range during configuration on the Extensions.

During the wizard, the user can quit the setup wizard at any time to start over with manual configuration. At the last step of the wizard, the user will be provided with summary for review, before the configuration is loaded. Once the setup is completed, the system is ready to go.

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Web GUI Configurations

There are eight main sections in the Web GUI for users to view the PBX status, configure and manage the

PBX.

System Status: Displays PBX status, System Status, Active calls and network status.

Extension/Trunk: To configure extensions, trunks and manage inbound/outbound call routes.

Call Features: To Configure call features such as ring groups, call queues, IVR ...Etc

PBX Settings: Extension ranges, SIP Settings, interfaces settings and MOH.

System Settings: To configure user management, network settings, firewall settings, change password, LDAP Server, HTTP Server, Email Settings, Time Settings, NTP server and login timeout.

Maintenance: To perform firmware upgrade, backup configurations, cleaner setup, reset/reboot, syslog setup and troubleshooting.

CDR: View call statistics and download CDR and call recordings.

Value-added Features: Zero Config, enable CTI server, CRM, PMS WebRTC.

Web GUI Languages

Currently the UCM6100 series web GUI supports English, Simplified Chinese, Traditional Chinese,

Spanish, French, Portuguese, Russian, Italian, Polish, German and etc.

Users can select the displayed language in web GUI login page, or at the upper right of the web GUI after logging in.

Figure 11: UCM6100 Web GUI Language

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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 Extension/TrunkExtensions.

2.

Click on "Add" to create a new extension. You will need User ID, Password and Voicemail Password information to register and use the extension later.

3.

Register the extension on your phone with the SIP User ID, SIP server and SIP Password information.

The SIP server address is the UCM6100 IP address.

4.

When your phone is registered with the extension, dial *97 to access the voicemail box. Enter the

Voicemail Password once you hear "Password" voice prompt.

5.

Once successfully logged in to the voicemail, you will be prompted with the Voice Mail Main menu.

6.

You are successfully connected to the PBX system now.

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SYSTEM SETTINGS

This section explains configurations for system-wide parameters on the UCM6100. System settings are under “Settings” tag on UCM6100 web GUI. System settings include User Management, Network Settings,

Firewall, Change Password, LDAP server, HTTP server, Email settings, Time Settings, NTP Server,

Recordings Storage and Login Timeout settings.

Network Settings

After successfully connecting the UCM6100 to the network for the first time, users could login the Web GUI and go to System SettingsNetwork 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System SettingsNetwork 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 6: UCM6102 Network SettingsBasic Settings

Method

MTU

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

The default setting is "Route".

Route

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

Switch

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

Dual

Both ports can be used for uplink connection. Users will need assign LAN 1 or

LAN 2 as the default interface in option "Default Interface" and configure

"Gateway IP" for this interface.

Specifies the Maximum Transmission Unit. (By default, it is 1500)

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IPv4 Address

Preferred DNS Server

Enter the preferred DNS server address. If Preferred DNS is used, UCM will try to use it as Primary DNS server.

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

IP Method

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

IP Address

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

Subnet Mask

Gateway IP

DNS Server 1

DNS Server 2

User Name

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.

Password

Layer 2 QoS

802.1Q/VLAN Tag

Enter the password to connect via PPPoE.

Assign the VLAN tag of the layer 2 QoS packets for WAN port.

The default value is 0.

Layer 2 QoS 802.1p

Priority Value

Assign the priority value of the layer 2 QoS packets for WAN port.

The default value is 0.

LAN (when Method is set to "Route")

IP Address

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

Subnet Mask

Enter the subnet mask. The default setting is 255.255.255.0.

DHCP Server Enable

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

DNS Server 1

DNS Server 2

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

Enter DNS server address 2. The default setting is 208.67.222.222.

Allow IP Address

From

Enter the DHCP IP Pool starting address. The default setting is 192.168.2.100.

Allow IP Address To

Enter the DHCP IP Pool ending address. The default setting is 192.168.2.254.

Default Gateway

Default IP Lease

Time

Layer 2 QoS

802.1Q/VLAN Tag

Layer 2 QoS 802.1p

Priority Value

Configure the default gateway assigned by the DHCP server.

Enter the IP lease time (in seconds). The default setting is 43200.

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 (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. The default setting is 255.255.0.0.

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Gateway IP

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

DNS Server 1

DNS Server 2

User Name

Password

Layer 2 QoS

802.1Q/VLAN Tag

Enter the DNS server 1 address for static IP settings. The default setting is 0.0.0.0.

Enter the DNS server 2 address for static IP settings.

Enter the user name to connect via PPPoE.

Enter the password to connect via PPPoE.

Assign the VLAN tag of the layer 2 QoS packets for LAN port.

The default value is 0.

Layer 2 QoS 802.1p

Priority Value

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

The default value is 0.

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

Default Interface

IP Method

IP Address

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) 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.

Subnet Mask

Gateway IP

DNS Server 1

DNS Server 2

User Name

Password

Layer 2 QoS

802.1Q/VLAN Tag

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

255.255.0.0.

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

Enter the DNS server 1 address for static IP settings.

The default setting is 0.0.0.0.

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.

Layer 2 QoS 802.1p

Priority Value

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

The default value is 0.

IPv6 Address

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

IP Method

Select Auto or Static. The default setting is Auto

IP Address

Enter the IP address for static IP settings.

IP Prefixlen

DNS Server 1

DNS Server 2

Enter the Prefix length for static settings. Default is 64

Enter the DNS server 1 address for static settings.

Enter the DNS server 2 address for static settings.

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LAN (when Method is set to "Route")

DHCP Server

Select Disable, Auto or DHCPv6.

Disable: the DHCPv6 server is disabled.

Auto: Stateless address auto configuration using NDP protocol.

DHCPv6: Stateful address auto configuration using DHCPv6 protocol.

DHCP Prefix

Enter DHCP prefix. (Default is 2001:db8:2:2::)

DHCP Prefixlen

Enter the Prefix length for static settings. Default is 64

DNS Server 1

Enter the DNS server 1 address for static settings.

Default is (2001:4860:4860::8888)

DNS Server 2

Enter the DNS server 2 address for static settings.

Default is (2001:4860:4860::8844)

Allow IP Address

From

Configure starting IP address assigned by the DHCP prefix and DHCP Prefixlen.

Allow IP Address To

Configure the ending IP address assigned by the DHCP Prefix and DHCP

Prefixlen.

Default IP Lease

Time

Configure the lease time (in seconds) of the IP address.

LAN (when Method is set to "Switch")

IP Method

Select Auto or Static. The default setting is Auto

IP Address

Enter the IP address for static IP settings.

IP Prefixlen

Default Interface

Enter the Prefix length for static settings. Default is 64

DNS Server 1

DNS Server 2

Enter the DNS server 1 address for static settings.

Enter the DNS server 2 address for static settings.

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

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.

IP Method

Select Auto or Static. The default setting is Auto

IP Address

IP Prefixlen

DNS Server 1

DNS Server 2

Enter the IP address for static IP settings.

Enter the Prefix length for static settings. Default is 64

Enter the DNS server 1 address for static settings.

Enter the DNS server 2 address for static settings.

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Table 7: UCM6104 Network SettingsBasic Settings

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

Switch

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

Method

IP Method

IP Address

Subnet Mask

Gateway IP

Dual

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

LAN 2 as the default interface in option "Default Interface" and configure

"Gateway IP" for this interface.

MTU

IPv4

Specifies the Maximum Transmission Unit. (By default, its 1500)

Preferred DNS Server

Enter the preferred DNS server address. If Preferred DNS is used, UCM will try to use it as Primary DNS server.

LAN (when Method is set to "Switch")

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. The default setting is 0.0.0.0.

DNS Server 1

DNS Server 2

User Name

Password

Layer 2 QoS

802.1Q/VLAN Tag

Enter the DNS server 1 address for static IP settings. The default setting is 0.0.0.0.

Enter the DNS server 2 address for static IP settings.

Enter the user name to connect via PPPoE.

Enter the password to connect via PPPoE.

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

0.

Layer 2 QoS 802.1p

Priority Value

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

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

Default Interface

IP Method

IP Address

Subnet Mask

Gateway IP

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

LAN 1 or LAN 2. The default interface is LAN 2.

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. The default setting is 0.0.0.0.

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DNS Server 1

DNS Server 2

User Name

Password

Layer 2 QoS

802.1Q/VLAN Tag

Enter the DNS server 1 address for static IP settings. The default setting is 0.0.0.0.

Enter the DNS server 2 address for static IP settings.

Enter the user name to connect via PPPoE.

Enter the password to connect via PPPoE.

Assign the VLAN tag of the layer 2 QoS packets for LAN port.

The default value is 0.

Layer 2 QoS 802.1p

Priority Value

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

IPv6 Address

LAN (when Method is set to "Switch")

IP Method

IP Address

Select Auto or Static. The default setting is Auto

Enter the IP address for static IP settings.

IP Prefixlen

DNS Server 1

Default Interface

Enter the Prefix length for static settings. Default is 64

Enter the DNS server 1 address for static settings.

DNS Server 2

Enter the DNS server 2 address for static settings.

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

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.

IP Method

IP Address

Select Auto or Static. The default setting is Auto

Enter the IP address for static IP settings.

IP Prefixlen

DNS Server 1

DNS Server 2

Enter the Prefix length for static settings. Default is 64

Enter the DNS server 1 address for static settings.

Enter the DNS server 2 address for static settings.

Table 8: UCM6108/UCM6116 Network SettingsBasic Settings

MTU

IPv4 Address

Specifies the Maximum Transmission Unit. (By default, it is 1500)

Preferred DNS Server

IP Method

IP Address

Subnet Mask

Gateway IP

DNS Server 1

DNS Server 2

Enter the preferred DNS server address. If Preferred DNS is used, UCM will try to use it as Primary DNS server.

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.

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User Name

Password

Layer 2 QoS

802.1Q/VLAN Tag

Layer 2 QoS 802.1p

Priority Value

IPv6 Address

IP Method

IP Address

IP Prefixlen

DNS Server 1

DNS Server 2

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.

Select Auto or Static. The default setting is Auto

Enter the IP address for static IP settings.

Enter the Prefix length for static settings. Default is 64

Enter the DNS server 1 address for static settings.

Enter the DNS server 2 address for static settings.

Method: Route

When the UCM6102 has method set to Route in network settings, WAN port interface is used for uplink connection and LAN port interface is used as a router. Please see a sample diagram below.

Figure 12: UCM6102 Network Interface Method: Route

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Method: Switch

For UCM6102, WAN port interface is used for uplink connection; LAN port interface is used as bridge for PC connection. For UCM6104, LAN 1 port interface is used for connection; LAN 2 port interface is used as bridge for PC connection.

Figure 13: UCM6102 Network Interface Method: Switch

Method: Dual

For UCM6102, both WAN port and LAN port are used for uplink connection. For UCM6104, WAN port will be mapped to LAN 1 interface; LAN port will be mapped to LAN 2 interface. Users will need assign

LAN 1 or LAN 2 as the default interface in option "Default Interface" and configure "Gateway IP" if static

IP is used for this interface.

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Figure 14: UCM6102 Network Interface Method: Dual

DHCP Client List

For UCM6102 that supports Route Mode, this feature can bind MAC to IP address on the LAN port.

When devices receive IP addresses from UCM6102 LAN port, they will be listed on the webUI under

System SettingsNetwork SettingsDHCP Client List” as shown below.

Figure 15: DHCP Client List

User can bind manually a MAC to an IP address by clicking on will pop up.

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Figure 16: Add MAC Address Bind

User needs to set the device MAC address and the IP that will be bound to it (the IP address needs to be within the UCM6102 DHCP range).

In order to bind a batch of listed MAC addresses, user needs to check first the MAC addresses to bind and click on addresses.

. A confirmation popup will be shown, click to bind the

Figure 17: Batch Add MAC Address Bind

After Clicking “OK” to confirm the binding, the “Bind Status” will change from “Unbind” to “Binding”.

802.1X

IEEE 802.1X is an IEEE standard for port-based network access control. It provides an authentication mechanism to device before the device is allowed to access Internet or other LAN resources. The UCM6100 supports 802.1X as a supplicant/client to be authenticated. The following diagram and figure show

UCM6100 uses 802.1X mode “EAP-MD5” on WAN port as client in the network to access Internet.

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Figure 18: UCM6100 Using 802.1X as Client

Figure 19: UCM6100 Using 802.1X EAP-MD5

The following table shows the configuration parameters for 802.1X on UCM6100. Identity and MD5 password are required for authentication, which should be provided by the network administrator obtained from the RADIUS server. If “EAP-TLS” or “EAP-PEAPv0/MSCHAPv2” is used as the 802.1X mode, users will also need upload 802.1X CA Certificate and 802.1X Client Certificate, which should be also generated from the RADIUS server.

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802.1X Mode

Identity

MD5 Password

802.1X Certificate

802.1X Client

Certificate

Static Routes

Table 9: UCM6100 Network Settings802.1X

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.

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

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

• Click on

to create a new IPv4 static route or click on to create a new IPv6 static route. The configuration parameters are listed in the table below.

• Once added, users can select

to edit the static route.

• Select

to delete the static route.

Destination

Netmask

Table 10: UCM6100 Network SettingsStatic Routes

Configure the destination IPv4 address or the destination IPv6 subnet for the

UCM6100 to reach using the static route.

Example:

IPv4 address - 192.168.66.4

IPv6 subnet - 2001:740:D::1/64

Configure the subnet mask for the above destination IPv4 address. If left blank, the default value is 255.255.255.255.

Example:

255.255.255.0

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Gateway

Configure the IPv4 or IPv6 gateway address so that the UCM6100 can reach the destination via this gateway. Gateway address is optional.

Example:

192.168.40.5 or 2001:740:D::1

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.

The following diagram shows a sample application of static route usage on UCM6104.

Figure 20: UCM6104 Static Route Sample

The network topology of the above diagram is as below:

• Network 192.168.69.0 has IP phones registered to UCM6104 LAN 1 address

• Network 192.168.40.0 has IP phones registered to UCM6104 LAN 2 address

• Network 192.168.66.0 has IP phones registered to UCM6104 via VPN

• Network 192.168.40.0 has VPN connection established with network 192.168.66.0

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In this network, by default the IP phones in network 192.168.69.0 are unable to call IP phones in network

192.168.66.0 when registered on different interfaces on the UCM6104. Therefore, we need configure a static route on the UCM6104 so that the phones in isolated networks can make calls between each other.

Figure 21: UCM6104 Static Route Configuration

Port Forwarding (UCM6102 only)

The UCM network interface supports router function which provides users the ability to do port forwarding.

If the UCM6102 LAN mode is set to "Route" under web GUISystem SettingsNetwork Settings page, port forwarding is available for configuration.

The port forwarding configuration is under web GUISystem SettingsNetwork SettingsPort

Forwarding page. Please see related settings in the table below.

Table 11: UCM6102 Network SettingsPort Forwarding

WAN Port

Specify the WAN port number or a range of WAN ports. Unlimited number of ports can be configured.

Note:

When it is set to a range, WAN port and LAN port must be configured with the same range, such as WAN port: 1000-1005 and LAN port: 1000-1005, and access from WAN port will be forwarded to the LAN port with the same port number, for example, WAN port 1000 will be port forwarding to LAN port 1000.

LAN IP

Specify the LAN IP address.

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LAN Port

Specify the LAN port number or a range of LAN ports.

Note:

When it is set to a range, WAN port and LAN port must be configured with the same range, such as WAN port: 1000-1005 and LAN port: 1000-1005, and access from WAN port will be forwarded to the LAN port with the same port number, for example, WAN port 1000 will be port forwarding to LAN port 1000.

Protocol Type

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

The following figures demonstrate a port forwarding example to provide phone’s Web GUI access to public side.

• The UCM6102 network mode is set to “Route”.

• The UCM6102 WAN port is connected to uplink switch, with a public IP address configured, e.g. 1.1.1.1.

• The UCM6102 LAN port provides DHCP pool that connects to multiple phone devices in the LAN network 192.168.2.x. The UCM6102 is used as a router, with gateway address 192.168.2.1.

• There is a GXP2160 connected under the LAN interface network of the UCM6102. It obtains IP address

192.168.2.100 from UCM6102 DHCP pool.

• On the UCM6102 Web GUISystem SettingsNetwork SettingsPort Forwarding, configure a port forwarding entry as the figures shows below.

• Click on

WAN Port: This is the port opened up on the WAN side for access purpose.

LAN IP: This is the GXP2160 IP address, under the LAN interface network of the UCM6102.

LAN Port: This is the port opened up on the GXP2160 side for access purpose.

Protocol Type: We select TCP here for Web GUI access using HTTP.

Figure 22: Create New Port Forwarding

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Figure 23: UCM6102 Port Forwarding Configuration

This will allow users to access the GXP2160 Web GUI from public side, by typing in the public IP address

(example: 1.1.1.1:8088).

Figure 24: GXP2160 Web Access Using UCM6102 Port Forwarding

OpenVPN

Open VPN settings allow the users to configure UCM6100 to use VPN features.

Table 12: UCM6100 System SettingsOpenVPN

Enable

Enable / Disable the open VPN feature.

Server Address

Configures the hostname/IP and port of the server. For example: 192.168.1.2:22

Server Protocol

Specify the protocol user, user should use the same settings as used on the server

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Device Mode

User

Compression

CA Cert

Client Cert

Client Key

Use the same setting as used on the server.

Dev TUN: Create a routed IP tunnel.

Dev TAP: Create an Ethernet tunnel.

Compress tunnel packets using the LZO algorithm on the VPN link. Don’t enable this unless it is also enabled in the server config file.

Upload as SSL/TLS root certificate. This file will be renamed as ‘ca.crt’ automatically.

Upload a client certificate. This file will be renamed as ‘cliend.crt’ automatically.

Upload a client private key. This file will be renamed as ‘client.key’ automatically.

Figure 25: OpenVPN feature on the UCM6100

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DDNS Settings

DDNS setting allows user to access UCM6100 via domain name instead of IP address.

The UCM supports DDNS service from the following DDNS provider:

• dydns.org

• noip.com

• freedns.afraid.org

• zoneedit.com

• oray.net

Here is an example of using noip.com for DDNS.

1.

Register domain in DDNS service provider. Please note the UCM6100 needs to have public IP access.

Figure 26: Register Domain Name on noip.com

2.

On Web GUISystem SettingsDDNS Settings, enable DDNS service and configure username, password and host name.

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Figure 27: UCM6100 DDNS Setting

3.

Now you can use domain name instead of IP address to connect to the UCM6100 Web GUI.

Figure 28: Using Domain Name to Connect to UCM6100

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Security Settings

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 the UCM6100, go to Web

GUISystem SettingsSecurity Settings page.

Static Defense

Under Web GUISystem SettingsSecurity SettingsStatic 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 FirewallStatic DefenseCurrent Service

Port

7777

Process

Asterisk

Type

TCP/IPv4

Protocol or Service

SIP

389

22

80

8089

Slapd

Dropbear

Lighthttpd

Lighthttpd

TCP/IPv4

TCP/IPv4

TCP/IPv4

TCP/IPv4

LDAP

SSH

HTTP

HTTPS

69

9090

6060

5060

4569

Opentftpd

Asterisk zero_config

Asterisk

Asterisk

UDP/IPv4

UDP/IPv4

UDP/IPv4

UDP/IPv4

UDP/IPv4

TFTP

SIP

UCM6100 zero_config service

SIP

SIP

5353

37435 zero_config

Syslogd

UDP/IPv4

UDP/IPv4

UCM6100 zero_config service

Syslog

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

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Ping Defense

Enable

SYN-Flood

Defense Enable

Ping-of-Death

Defense Enable

Table 14: Typical Firewall Settings

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.

Allows the UCM6100 to handle excessive amounts of SYN packets from one source and keep the web portal accessible. There are two options available and only one of these options may be enabled at one time.

• eth(0)LAN defends against attacks directed to the LAN IP address of the

UCM6100.

• eth(1)WAN defends against attacks directed to the WAN IP address of the

UCM6100.

SYN Flood Defense will limit the amount of SYN packets accepted by the UCM from one source to 10 packets per second. Any excess packets from that source will be discarded.

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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Rule Name

Action

Type

Service

Figure 29: Create New Firewall Rule

Table 15: Firewall Rule Settings

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".

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Save the change and click on "Apply" button. Then submit the configuration by clicking on "Apply Changes" on the upper right of the web page. The new rule will be listed at the bottom of the page with sequence number, rule name, action, protocol, type, source, destination and operation. More operations below:

• Click on

to edit the rule

• Click on

to delete the rule

Dynamic Defense

Dynamic defense is supported on the UCM6100 series. It can blacklist hosts dynamically when the LAN mode is set to "Route" under web GUISystem SettingsNetwork SettingsBasic Settings page. If enabled, the traffic coming into the UCM6100 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 UCM6100 firewall, which will then be displayed in the web page. Please refer to the following table for dynamic defense options on the UCM6100.

Table 16: UCM6100 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.

Allowed IPs and ports range, multiple IP addresses and port range.

For example

192.168.5.100-

192.168.5.200 1500:2000

The following figure shows a configuration example like this:

• If a host at IP address 192.168.5.7 initiates more than 20 TCP connections to the UCM6100 within 1 minute, it will be added into UCM6100 blacklist.

• This host 192.168.5.7 will be blocked by the UCM6100 for 5000 seconds.

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• Since IP range 192.168.5.100-192.168.5.200 is in whitelist, if a host initiates more than 20 TCP connections to the UCM6100 within 1 minute, it will not be added into UCM6100 blacklist. It can still establish TCP connection with the UCM6100.

Figure 30: Configure Dynamic Defense

Fail2ban

Fail2Ban feature on the UCM6100 provides intrusion detection and prevention for authentication errors in

SIP REGISTER, INVITE and SUBSCRIBE. Once the entry is detected within "Max Retry Duration", the

UCM6100 will take action to forbid the host for certain period as defined in "Banned Duration". This feature helps prevent SIP brute force attacks to the PBX system.

Fail2ban has also an interesting feature which allows the detection of login attacks against the web GUI of the UCM and de the necessary operations to block the attackers, this can be configured and set using a

Max Retry period after which is an attacker failed the UCM will block their IP, and they will be banned from accessing the web GUI of the UCM.

Both above features can be enabled and configured under the local settings tab of fail2ban security feature, as for the ports, they default to the listening ports of SIP and HTTP used by the UCM.

The Following figure shows the settings available to configure and tune fail2ban defense.

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Figure 31: Fail2ban Settings

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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 600. If set to 0, 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 600.

Configure the number of authentication failures during "Max Retry Duration" before the host is banned. The default setting is 5.

Configure IP address, CIDR mask or DNS host in the whitelist. 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

Listening Port

Number

MaxRetry

Login Attack Defense

Listening Port

Number

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. By default, port 5060 will be used for UDP and TCP, and port 5061 will be used for TLS.

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".

Enables defense against excessive login attacks to the UCM’s web GUI.

The default setting is disabled.

This is the Web GUI listening port number which is configured under System

SettingsHTTP ServerPort. The default is 8089.

When the number of failed login attempts from an IP address exceeds the

MaxRetry number, that IP address will be banned from accessing the UCM Web

UI.

Blacklist

Blacklist

Users will be able to view the IPs that have been blocked by UCM.

SSH Access

SSH switch now is available via Web GUI and LCD. User can enable or disable SSH access directly from

Web GUI or LCD screen. For web SSH access, please log in UCM 6100 web interface and go to System

SettingsSecurity SettingsSSH Access. By default, SSH access is disabled for security concerns. It is highly recommended to only enable SSH access for debugging purpose.

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Figure 32: SSH Access

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".

Term Explanation:

cn= Common Name ou= Organization Unit dc= Domain Component

These are all parts of the LDAP data Interchange Format, according to RFC 2849, which is how the LDAP tree is filtered.

If users have the Grandstream phone provisioned by the UCM6100, the LDAP directory will be 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.

UCM can also act as a LDAP client to download phonebook entries from another LDAP server.

To access LDAP server and client settings, go to Web GUISettingsLDAP Server.

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LDAP Server Configurations

The following figure shows the default LDAP server configurations on the UCM6100.

Figure 33: 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.

After that users can view the local LDAP entries for the local users but to change the information for each user, this can be done under user management tab at the maintenance Tab.

Figure 34: Default LDAP Phonebook DN

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Figure 35: Default LDAP Phonebook Attributes

LDAP Phonebook

Users could use the default phonebook, edit the default phonebook, add new phonebook, import phonebook on the LDAP server as well as export phonebook from the LDAP server. The first phonebook with default phonebook dn "ou=pbx,dc=pbx,dc=com" displayed on the LDAP server page is for extensions in this PBX. Users cannot add or delete contacts directly. The contacts information will need to be modified via Web GUIExtension/TrunkExtensions first. The default LDAP phonebook will then be updated automatically.

Figure 36: LDAP ServerLDAP Phonebook

Add new phonebook

A new sibling phonebook of the default PBX phonebook can be added by clicking on "Add" under "LDAP

Phonebook" section.

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Figure 37: Add LDAP Phonebook

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

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

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

Figure 38: Edit LDAP Phonebook

Import phonebook from your computer to LDAP server

Click on “Import Phonebook” and a dialog will prompt as shown in the figure below.

Figure 39: Import Phonebook

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The file to be imported must be a CSV file with UTF-8 encoding. Users can open the CSV file with Notepad and save it with UTF-8 encoding.

Here is how a sample file looks like. Please note “Account Number” and “Phonebook DN” fields are required.

Users could export a phonebook file from the UCM6100 LDAP phonebook section first and use it as a sample to start with.

Figure 40: Phonebook CSV File Format

The Phonebook DN field is the same “Phonebook Prefix” entry as when the user clicks on “Add” to create a new phonebook. Therefore, if the user enters “phonebook” in “Phonebook DN” field in the CSV file, the actual phonebook DN “ou=phonebook,dc=pbx,dc=com” will be automatically created by the UCM6100 once the CSV file is imported.

In the CSV file, users can specify different phonebook DN fields for different contacts. If the phonebook DN already exists on the UCM6100 LDAP Phonebook, the contacts in the CSV file will be added into the existing phonebook. If the phonebook DN doesn’t exist on the UCM6100 LDAP Phonebook, a new phonebook with this phonebook DN will be created.

The sample phonebook CSV file in above picture will result in the following LDAP phonebook in the

UCM6100.

Figure 41: LDAP Phonebook After Import

As the default LDAP phonebook with DN “ou=pbx,dc=pbx,dc=com” cannot be edited or deleted in LDAP phonebook section, users cannot import contacts with Phonebook DN field “pbx” if existed in the CSV file.

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Export phonebook to your computer from UCM6100 LDAP server

Select the checkbox for the LDAP phonebook and then click on “Export Selected Phonebook” to export the selected phonebook. The exported phonebook can be used as a record or a sample CSV file for the users to add more contacts in it and import to the UCM6100 again.

LDAP Client Configurations

Figure 42: Export Selected LDAP Phonebook

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):

Server Address: LDAP server IP address

Base DN: dc=pbx,dc=com

User Name: cn= “LDAP server login name”, dc=pbx, dc=com [matching LDAP server format]

Password: “LDAP server login password”

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

Port: 389

The following figure gives a sample configuration for UCM6100 acting as a LDAP client.

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Figure 43: LDAP Client Configurations

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

Server Address: The IP address or domain name of the 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 GXP2170 to successfully use

the LDAP server as configured in

Figure 33: LDAP Server Configurations

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Figure 44: GXP21XX LDAP Phonebook Configuration

HTTP Server

The UCM6100 embedded web server responds to HTTP/HTTPS GET/POST requests. Embedded HTML pages allow the users to configure the PBX through a Web browser such as Microsoft IE, Mozilla Firefox and Google Chrome. By default, the PBX can be accessed via HTTPS using Port 8089 (e.g., https://192.168.40.50:8089 ). Users could also change the access protocol and port as preferred under Web

GUISystem SettingsHTTP Server.

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Redirect From Port 80

Protocol Type

Port

Enable IP whitelist

Permitted IP(s)

TLS Private Key

TLS Cert

Table 18: HTTP Server Settings

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

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

Select HTTP or HTTPS. The default setting is "HTTPS". This is also the protocol used for zero config when the end point device downloads the config file from the UCM6100.

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

If enabled, only the IP address on the permitted IP list will be allowed to access the UCM’s web GUI.

Add an IP address to the list of allowed IPs to access UCM’s web GUI.

Ex: 192.168.6.233 / 255.255.255.255

Upload private key for the built-in http server

Upload certificate for the built-in http server and override the existing one.

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

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System SettingsEmail Settings.

Table 19: Email Settings

TLS Enable

Type

Domain

Server

Username

Password

Enable or disable TLS during transferring/submitting your Email to another

SMTP server. The default setting is "Yes".

Select Email type.

MTA: Mail Transfer Agent. The Email will be sent from the configured domain. When MTA is selected, there is no need to set up SMTP server for it or no user login is required. However, the Emails sent from

MTA might be considered as spam by the target SMTP server.

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

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

Specify the SMTP server when using type "Client".

Username is required when using type "Client". Often is the Email address.

Password to login for the above Username (Email address) is required

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Display Name

Sender

when using type "Client".

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

Specify the sender's Email address.

For example: [email protected]

The following figure shows a sample Email setting on the UCM6100, assuming the Email is using

smtp.gmail.com as the SMTP server.

Figure 45: 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.

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Email Templates

The Email templates on the UCM6100 can be used for email notification the configuration parameters can be accessed via Web GUISystem SettingsEmail SettingsEmail Templates.

Figure 46: Email Templates

To configure the email template, simply click the button under Options column, and edit the template.

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-

-

Click on

Figure 47: Conference Schedule Template

Users have the ability to preview mail sample by clicking on .

in order to restore the default email template.

-

Finally, users can click on to upload a custom picture to the email template to display their own logo in the sent mails for example

Email Send Log

Under UCM Web GUISystem SettingsEmail SettingsEmail Send Log, users could search, filter and check whether the email is sent out successfully or not. This page will also display the corresponding error message if the email is not sent out successfully.

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Figure 48: Email Send log

Field

Start Time

End Time

Receivers

Send Result

Return Code

Email Send Module

Table 20: Email Log

Description

Enter the start time for filter

Enter the end time for filter

Enter the email recipient, while searching for multiple recipients, please separate then with comma and no spaces.

Enter the status of the send result to filter with

Enter the email code to filter with

Select the email module to filter with from the drop-down list, which contains:

All Modules

Extension

Voicemail

Conference Schedule

User Password

Alert Events

CDR

Fax

Test

Email logs will be shown on bottom of the “Email Send Log” page, as shown on the following figure.

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Figure 49: Email Logs

Time Settings

Auto Time Updating

The current system time on the UCM6100 can be found under Web GUISystem

StatusDashboardPBX Status.

To configure the UCM6100 to update time automatically, go to Web GUISystem SettingsTime

SettingsAuto Time Updating.

--------------------------------------------------------------------------------------------------------------------------------------------

Note:

The configurations under Web GUISystem SettingsTime SettingsAuto Time Updating page require reboot to take effect. Please consider configuring auto time updating related changes when setting up the UCM6100 for the first time to avoid service interrupt after installation and deployment in production.

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Table 21: Time Auto Updating

Remote NTP Server

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.

Enable DHCP Option 2

Enable DHCP Option 42

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 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".

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Time Zone

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.

Self-Defined Time Zone

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

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

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

MTZ+6MDT+5

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

M4.1.0,M11.1.0

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

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

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

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

,Sat).

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

April to the 1st Sunday of November.

Set Time Manually

To manually set the time on the UCM6100, go to Web GUISystem SettingsTime SettingsSet Time

Manually. The format is YYYY-MM-DD HH:MI:SS.

Figure 50: Set Time Manually

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Note:

Manually setup time will take effect immediately after saving and applying change in the Web GUI. 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System

SettingsTime SettingsAuto Time 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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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System

SettingsTime SettingsNTP 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.

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System SettingsTime SettingsOffice Time. Press "Create New Office Time" to create an office time.

Figure 51: Create New Office Time

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Start Time

End Time

Week

Show Advanced Options

Table 22: 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 option 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.

Month

Day

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.

• Click on

Figure 52: System SettingsTime SettingsOffice Time

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System SettingsTime SettingsHoliday. Click on "Create New Holiday" to create holiday time.

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Figure 53: Create New Holiday

Name

Holiday Memo

Month

Day

Show Advanced Options

Table 23: Create New Holiday

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.

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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• Click on

Figure 54: System SettingsTime SettingsHoliday

to edit the holiday.

• Click on

to delete the holiday.

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

----------------------------------------------------------------------------------------------------------------------------- ---------------

Note:

For more details on how to use office time and holiday, please refer to the link below: http://www.grandstream.com/sites/default/files/Resources/office_time_and_holiday_on_ucm6xxx.pdf

--------------------------------------------------------------------------------------------------------------------------------------------

Recordings Storage

The UCM6100 supports call recordings automatically or manually and the recording files can be saved in external storage plugged in the UCM6100 or on the UCM6100 locally. To manage the recording storage, users can go to UCM6100 web GUIPBX SettingsRecordings Storage page and select whether to store the recording files in USB Disk, SD card or locally on the UCM6100.

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Figure 55: PBX SettingsRecordings Storage

• If “Enable Auto Change” is selected, the recording files will be automatically saved in the available

USB Disk or SD card plugged into the UCM6100. If both USB Disk and SD card are plugged in, the recording files will be always saved in the USB Disk.

• If “Local” is selected, the recordings will be stored in UCM6100 internal storage.

• If “USB Disk” or “SD Card” is selected, the recordings will be stored in the corresponding plugged in external storage device. Please note the options “USB Disk” and “SD Card” will be displayed only if they are plugged into the UCM6100.

Once “USB Disk” or “SD Card” is selected, click on “OK”. The user will be prompted to confirm to copy the local files to the external storage device.

Figure 56: Recordings Storage Prompt Information

Click on “OK” to continue. The users will be prompted a new dialog to select the categories for the files to be copied over.

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Figure 57: Recording Storage Category

On the UCM6100, recording files are generated and exist in 3 categories: normal call recording files, conference recording files, and call queue recording files. Therefore, users have the following options when select the categories to copy the files to the external device:

Recording Files: Copy the normal recording files to the external device.

Conference: Copy the conference recording files to the external device.

Queue: Copy the call queue recording files to the external device.

All: Copy all recording files to the external device.

Google Service Settings Support

UCM6100 now supports Google OAuth 2.0 authentication. This feature is used for supporting UCM6100 conference scheduling system. Once OAuth 2.0 is enabled, UCM6100 conference system can access

Google calendar to schedule or update conference.

Google Service Settings can be found under Web GUICall FeaturesConferenceGoogle Service

SettingsGoogle Service Settings.

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Figure 58: Google Service SettingsOAuth2.0 Authentication

If you already have OAuth2.0 project set up on Google Developers web page, please use your existing login credential for “OAuth2.0 Client ID” and “OAuth2.0 Client Secret” in the above figure for the UCM6100 to access Google Service.

If you do not have OAuth2.0 project set up yet, please following the steps below to create new project and obtain credentials:

1.

Go to Google Developers page https://console.developers.google.com/start Create a New Project in

Google Developers page.

Figure 59: Google ServiceNew Project

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2.

Enable Calendar API from API Library.

3.

Click “Credentials” on the left drop down menu to create new OAuth2.0 login credentials.

Figure 60: Google ServiceCreate New Credential

4.

Use the newly created login credential to fill in “OAuth2.0 Client ID” and “OAuth2.0 Client Secret”.

5.

Click “Get Authentication Code” to obtain authentication code from Google Service.

Figure 61: Google ServiceOAuth2.0 Login

6.

Now UCM6100 is connected with Google Service.

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PROVISIONING

Overview

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

TFTP/HTTP/HTTPS download. All Grandstream SIP devices support a proprietary binary format configuration file and XML format configuration file. The 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Value-added FeaturesZero 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 62: 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 63: 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Value-added FeaturesZero ConfigZero

Config Settings, fill in the auto provision information.

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Figure 64: Auto Provision Settings

Enable Zero Config

Enable Automatic

Configuration Assignment

Auto Assign Extension

Table 24: Auto Provision Settings

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.

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.

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Zero Config Extension

Segment

Enable Pick Extension

Pick Extension Segment

Pick Extension Period (hour)

Subnet Whitelist

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 GUIPBX SettingsGeneral SettingsGeneral pageExtension Preference section: "Auto Provision Extensions".

If enabled, the extension list will be sent out to the device after receiving the device's request. This feature is for the GXP series phones that support selecting extension to be provisioned via phone's LCD. The default setting is disabled.

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 GUIPBX SettingsGeneral

SettingsGeneral pageExtension Preference section: "Pick

Extensions".

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

This feature allows the UCM to provision devices in different subnets other than UCM network.

Enter subnets IP addresses to allow devices within these subnets to be provisioned. The syntax is <IP>/<CIDR>.

Examples:

10.0.0.1/8

192.168.6.0/24

Note: Only private IP ranges (10.0.0.0 | 172.16.0.0 | 192.168.0.0) are supported.

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

Grandstream endpoints are automatically discovered after bootup. Users could also manually discover 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)

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Click on "Auto Discover" under Web GUIValue-added FeaturesZero ConfigZero Config, fill in the

"Scan Method" and "Scan IP". The IP address segment will be automatically filled in based on the network mask detected on the UCM6100. If users need scan the entire network segment, enter 255 (for example,

192.168.5.255) instead of a specific IP address. Then click on "Save" to start discovering the devices within the same network. To successfully discover the devices, "Zero Config" needs to be enabled on the

UCM6100 web GUIValue-added FeaturesZero ConfigZero Config Settings.

Figure 65: Auto Discover

The following figure shows a list of discovered phones. The MAC address, IP Address, Extension (if assigned), Version, Vendor, Model, Connection Status, Create Config, Options (Edit /Delete /Update

/Reboot /Access Device WebGUI) are displayed in the list.

Figure 66: Discovered Devices

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Uploading Devices List

Besides the built-in discovery method on the UCM, users could prepare a list of devices on .CSV file and upload it by clicking on the button , after which a success message prompt should be displayed.

Users need to make sure that the CSV file respects the format as shown on the following figure and that the entered information is correct (valid IP address, valid MAC address, device model ..etc), otherwise the UCM will reject the file and the operation will fail:

Figure 67: Device list - CSV file sample

Managing discovered devices:

Sorting: Press

or to sort per MAC Address, IP Address, Version, Vendor, Model or Create

Config columns from lower to higher or higher to lower respectively.

Filter: Select a filter

All: Display all discovered devices. to display corresponding results.

Scan Results: Display only manually discovered devices. [Discovery]

IP Address: Enter device IP and press Search button.

MAC Address: Enter device MAC and press Search button.

Model: Enter a model name and press Search button. Example: GXP2130.

Figure 68: Managing Discovered Devices

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From the main menu of zero config, users can perform the following operations:

• Click on

in order to access to the discovery menu as shown on [Discovery] section.

• Click on

• Click on

• Click on

to add a new device to zero config database using its MAC address.

to delete selected devices from the zero config database.

to modify selected devices.

• Click on

to batch update a list of devices, the UCM on this case will send SIP NOTIFY message to all selected devices in order to update them at once.

• Click on

to reboot selected devices (the selected devices, should have been provisioned with extensions since the phone will authenticate the server which is trying to send it reboot command).

• Click on

to clear all devices configurations.

• Click on

to upload CSV file containing list of devices.

All these operations will be detailed on the next sections.

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 GUIValue-added FeaturesZero ConfigGlobal Policy

• Web GUIValue-added FeaturesZero ConfigGlobal Templates.

Global Templates configuration has higher priority to Global Policy configuration.

Global Policy can be accessed in web GUIValue-added FeaturesZero ConfigGlobal Policy page.

On the top of the configuration table, users can select category in the "Options" dropdown list to quickly navigate to the category. The categories are:

Localization: configure display language, data and time.

Phone Settings: configure dial plan, call features, NAT, call progress tones and etc.

Contact List: configure LDAP and XML phonebook download.

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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.

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

Figure 69: Global Policy Categories

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

Table 25: Global Policy ParametersLocalization

Language settings

Language

Date and Time

Date Format

Select the LCD display language on the SIP end device.

Time Format

NTP Server

Time Zone

Configure the date display format on the SIP end device’s LCD.

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

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

Configure the time zone used on the SIP end device.

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Default Call Settings

Dial Plan

Enable Call Features

Use # as Dial Key

Auto Answer by Call-info

NAT Traversal

Use Random Port

General Settings

Call Progress Tones

Table 26: Global Policy ParametersPhone Settings

Configure the default dial plan rule. For syntax and examples, please refer to user manual of the SIP devices to be provisioned for more details.

When enabled, “Do Not Disturb”, “Call Forward” and other call features can be used via the local feature code on the phone. Otherwise, the ITSP feature code will be used.

If set to “Yes”, pressing the number key “#” will immediately dial out the input digits.

If set to "Yes", the phone will automatically turn on the speaker phone to answer incoming calls after a short reminding beep, based on the SIP

Call-Info header sent from the server/proxy. Default setting is enabled.

Configure which NAT traversal mechanism will be enabled on the endpoint device.

If set to "STUN" and STUN server is configured, the phone system will periodically send STUN message to the SUTN server to get the public IP address of its NAT environment and keep the NAT port open. STUN will not work if the NAT is symmetric type.

If set to “Keep-alive”, the phone system will send the STUN packets to maintain the connection that is first established during registration of the phone. The “Keep-alive” packets will fool the NAT device into keeping the connection open and this allows the host server to send SIP requests directly to the registered phone.

If it needs to use OpenVPN to connect host server, it needs to set it to

“VPN”.

If the firewall and the SIP device behind the firewall are both able to use

UPNP, it can be set to “UPNP”. The both parties will negotiate to use which port to allow SIP through.

The default setting is "Keep-alive".

Configure whether to allow the endpoint device to use random ports for both SIP and RTP messages. This is usually necessary when multiple phones are behind the same full cone NAT. The default setting is "No".

Note: This parameter must be set to "No" for Direct IP Calling to work.

Configure call progress tones including ring tone, dial tone, second dial tone, message waiting tone, ring back tone, call waiting tone, busy tone and reorder tone using the following syntax: f1=val, f2=val[, c=on1/ off1[- on2/ off2[- on3/ off3]]];

• Frequencies are in Hz and cadence on and off are in 10ms).

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HEADSET Key Mode

LDAP Phonebook

Source

Address

Port

Base DN

User Name

Password

Number Filter

Name Filter

Version

Name Attribute

Number Attribute

Display Name

• “on” is the period (in ms) of ringing while “off” is the period of silence.

Up to three cadences are supported.

• Please refer to user manual of the SIP devices to be provisioned for more details

Select “Default Mode” or “Toggle Headset/Speaker” for the Headset key.

Please refer to user manual of the SIP devices provisioned for details.

Table 27: Global Policy ParametersContact 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 UCM6100

Web GUISystem SettingsLDAP 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 begins. 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.

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

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Max Hits

Search Timeout

Sort Results

Incoming Calls

Outgoing Calls

Lookup Display Name

XML Phonebook

Phonebook XML Server

Phonebook Download

Interval

Remove manually-edited entries on download

can be displayed.

Example:

%cn %sn %telephoneNumber

Specify the maximum number of results to be returned by the LDAP server.

Valid range is 1 to 3000. The default value is 50.

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

Specify whether the searching result is sorted or not. 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.

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.

Upgrade and Provision

Firmware Source

Table 28: Global Policy ParametersMaintenance

Firmware source via ZeroConfig provisioning could a URL for external server address, local UCM directory or USB media if plugged in to the

UCM6100.

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Upgrade via

Server Path

File Prefix

File Postfix

Allow DHCP Option 43/66

Automatic Upgrade

Select a source to get the firmware file:

URL

If select to use URL to upgrade, complete the configuration for the following four parameters: “Upgrade Via”, “Server Path”, “File Prefix” and “File Postfix”.

Local UCM Server

Firmware can be uploaded to the UCM6100 internal storage for firmware upgrade. If selected, click on “Manage Storage” icon next to

“Directory” option, upload firmware file and select directory for the end device to retrieve the firmware file.

Local USB Media

If selected, the USB storage device needs to be plugged into the

UCM6100 and the firmware file must be put under a folder named

“ZC_firmware” in the USB storage root directory.

Local SD Card Media

If selected, an SD card needs to be plugged into the UCM6100 and the firmware file must be put under a folder named “ZC_firmware” in the USB storage root directory.

When URL is selected as firmware source, configure upgrade via TFTP,

HTTP or HTTPS.

When URL is selected as firmware source, configure the firmware upgrading server path.

When URL is selected as firmware source, configure the firmware file prefix. If configured, only the firmware with the matching encrypted prefix will be downloaded and flashed into the phone, if URL is selected.

When URL is selected as firmware source, configure the firmware file postfix. 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, the endpoint device will automatically upgrade if a new firmware is detected. Users can select automatic upgrading by day, by week or by minute.

By week

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

By day

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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.

Firmware Upgrade Rule

Web Access

Admin Password

End-User Password

Web Access Mode

Web Server Port

Security

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.

Disable Telnet/SSH

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.

Syslog

Syslog Server

Syslog Level

Send SIP Log

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.

Table 29: Global Policy ParametersNetwork 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

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 non-

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symmetric 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.

Keep Alive Interval

Wallpaper

Screen Resolution

1024 x 600

Screen Resolution

800 x 400

Screen Resolution

480 x 272

Screen Resolution

320 x 240

Table 30: Global Policy ParametersCustomization

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

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.

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Global Templates

Global Templates can be accessed in web GUIValue-added FeaturesZero ConfigGlobal

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

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

Template Name

Description

Active

Table 31: 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.

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Figure 70: Edit Global Template

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

Click on “Save” to save this global template.

• The created global templates will show in the Web GUIValue-added FeaturesZero

ConfigGlobal Templates page. Users can click on to delete the global template or click on

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

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

Model Configuration

Model Templates

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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 Value-added FeaturesZero

ConfigModel 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 32: Create New Model Template

Model

Template Name

Description

Default Model Template

Active

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.

• Click on

to edit the model template.

The editing window for model template is shown in the following figure. In the “Options” field, enter the option name key word, the option that contains the key word will be listed. User could then select the option and click on “Add Option” to add it into the model template.

Once added, the option will be shown in the list below. On the left side of each option, users can click on

to remove this option from the model template. On the right side of each option, users can click on to reset the option to the default value.

User could also click on “Add New Field” to add a P value number and the value to the configuration. The following figure shows setting P value “P1362” to “en”, which means the display language on the LCD is set to English. For P value information of different models, please refer to configuration template here

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http://www.grandstream.com/support/tools.

Figure 71: Edit Model Template

• Click on Save when done. The model template will be displayed on Web GUIValue-added

FeaturesZero ConfigModel 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.

Model Update

UCM6100 zero config feature supports provisioning all models of Grandstream SIP end devices. Templates for most of the Grandstream models are built in with the UCM6100 already. Templates for GS Wave and

Grandstream surveillance products require users to download and install under Web GUIValue-added

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FeaturesZero ConfigModel Update first before they are available in the UCM6100 for selection. After downloading and installing the model template to the UCM6100, it will show in the dropdown list for “Model” selection when editing the model template.

• Click on

to download the template.

• Click on

to upgrade the model template. Users will see this icon available if the device model has template updated in the UCM6100.

Figure 72: Template Management

In case the UCM6100 is placed in the private network and Internet access is restricted, users will not be able to get packages by downloading and installing from the remote server. Model template package can be manually uploaded from local device through Web GUI. Please contact Grandstream customer support if the model package is needed for manual uploading.

Figure 73: Upload Model Template Manually

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Device Configuration

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

Create New Device

Besides configuring the device after the device is discovered, users could also directly create a new device and configure basic settings before the device is discovered by the UCM6100. Once the device is plugged in, it can then be discovered and provisioned. This gives the system administrator adequate time to set up each device beforehand.

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

1.

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

2.

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

3.

Click on “Create New Device” to save the configuration for this device.

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Figure 74: Create New Device

Manage Devices

The device manually created or discovered from Auto Discover will be listed in the Web GUIValue-added

FeaturesZero ConfigZero Config page. Users can see the devices with their MAC address, IP address, vendor, model and etc.

Figure 75: Manage Devices

• Click on

to access the Web GUI of the phone.

• Click on

to edit the device configuration.

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A new dialog will be displayed for the users to configure “Basic” settings and “Advanced” settings. “Basic” settings have the same configurations as displayed when manually creating a new device, i.e., account, line key and MPK settings; “Advanced” settings allow users to configure more details in a five-level structure.

Figure 76: Edit Device

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

(1)

Global Policy

This is the lowest level configuration. The global policy configured in Web GUIValue-added

FeaturesZero ConfigGlobal Policy will be applied here. Clicking on “Modify Global Policy” to redirect to page Value-added FeaturesZero ConfigGlobal Policy.

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(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 GUIValue-added FeaturesZero ConfigModel

Templates page. Please see default model template option in

[Table 32: Create New Model

Template].

(4)

Model Templates

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

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

(5)

Customize Device Settings

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

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Figure 77: Edit Customize Device Settings

Scroll down in the dialog to view and edit the device-specific options. If the users would like to add more options which are not in the pre-defined list, click on “Add New Field” to add a P value number and the value to the configuration. The following figure shows setting P value “P1362” to “en”, which means the display language on the LCD is set to English. The warning information on right tells that the option matching the P value number exists and clicking on it will lead to the matching option.

For P value information of different models, please refer to configuration template here http://www.grandstream.com/sites/default/files/Resources/config-template.zip

.

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Figure 78: Add P Value in Customize Device Settings

• Select multiple devices that need to be modified and then click on to batch modify devices.

If selected devices are of the same model, the configuration dialog is like the following figure. Configurations in five levels are all available for users to modify.

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Figure 79: Modify Selected Devices - Same Model

If selected devices are of different models, the configuration dialog is like the following figure. Click on

to view more devices of other models. Users are only allowed to make modifications in Global

Templates and Global Policy level.

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Figure 80: Modify Selected Devices - Different Models

--------------------------------------------------------------------------------------------------------------------------------------------

Note:

Performing batch operation will override all the existing device configuration on the page.

--------------------------------------------------------------------------------------------------------------------------------------------

After the above configurations, save the changes and go back to Web GUIValue-added FeaturesZero

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

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Figure 81: Device List in Zero Config

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

Sample Application

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

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

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

1.

Go to Web GUIValue-added FeaturesZero ConfigZero Config Settings, select “Enable Zero

Config”.

2.

Go to Web GUIValue-added FeaturesZero ConfigGlobal Policy, configure Date Format, Time

Format and Firmware Source as follows.

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Figure 82: Zero Config Sample - Global Policy

3.

Go to web GUIValue-added FeaturesZero ConfigModel 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Value-added FeaturesZero ConfigModel 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 GUIValue-added

FeaturesZero ConfigZero Config.

6.

On web GUIValue-added FeaturesZero ConfigZero Config page, users could identify the devices by their MAC addresses or IP addresses displayed on the list. Click on to edit the device settings.

7.

For each of the 5 phones used by English speaking customer support, in “Basic” settings select an available extension for account 1 and click on “Save”. Then click on “Advanced” settings tab to bring up the following dialog. Users will see the English support template is applied since this is the default model template. A preview of the device settings will be listed on the right side.

Figure 83: Zero Config Sample - Device Preview 1

8.

For the 3 phones used by Spanish support, in “Basic” settings select an available extension for account

1 and click on “Save”. Then click on “Advanced” settings tab to bring up the following dialog.

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Figure 84: Zero Config Sample - Device Preview 2

Select “Spanish Support Template” in

Model Template”. The preview of the device settings is displayed on the right side and we can see the language is set to “Español” since Model Template has the higher priority for the option “Language”, which overrides the value configured in default model template.

9.

For the GXV3275 used by the customer support supervisor, select an available extension for account

1 on “Basic” settings and click on “Save”. Users can see the preview of the device configuration in

“Advanced” settings. There is no model template configured for GXP2170.

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Figure 85: Zero Config Sample - Device Preview 3

10.

Click on “Apply Changes” to apply saved changes.

11.

On the Web GUIValue-added FeaturesZero ConfigZero 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Extension/TrunkExtensions. Click on "Add" and a new dialog window will show for users to fill in the extension information.

Figure 86: Create New Device

SIP extension options are divided into four categories:

• Basic Settings

• Media

• Features

• Specific Time

• Follow me

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Select first which type of extension: SIP Extension, IAX Extension or FXS Extension.

The configuration parameters are as follows.

Table 33: SIP Extension Configuration ParametersBasic Settings

General

Extension

CallerID Number

Permission

SIP/IAX Password

Auth ID

Enable Voicemail

Voicemail Password

Skip Voicemail Password

Verification

Send Voicemail to Email

Keep Voicemail after

Emailing

Enable Keep-alive

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.

Configure the authentication ID for the user. If not configured, the extension number will be used for authentication.

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

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

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.

Allows users to send voicemail recording as .wav file attachment to specified email addresses. Default setting is “No”.

Note: When set to “Default”, the global settings in Call Features

VoicemailVoicemail Email Settings will be used.

Whether to keep the local voicemail recording after sending them. If set to

“Default”, the global settings will be used.

Note: When set to “Default”, the global settings in Call Features

VoicemailVoicemail Email Settings will be used.

If enabled, empty SDP packet will be sent to the SIP server periodically to keep the NAT port open. The default setting is "Yes".

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Keep-alive Frequency

Disable This Extension

User Settings

First Name

Last Name

Email Address

User Password

Language

Concurrent Registrations

Mobile Phone Number

Configure the Keep-alive interval (in seconds) to check if the host is up.

The default setting is 60 seconds.

If selected, this extension will be disabled on the UCM6100.

Note:

The disabled extension still exists on the PBX but cannot be used.

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

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

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

Configure the password for user portal access. A random numeric password is automatically generated. It is recommended to use the randomly generated password for security purpose.

Select the voice prompt language to be used for this extension. The default setting is "Default" which is the selected voice prompt language under web

GUIPBX SettingsVoice PromptLanguage Settings. 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

SettingsVoice PromptLanguage Settings.

The maximum endpoints which can be registered into this extension. For security concerns, the default value is 1.

Configure the phone number for the extension, user can type the related star code for phone number followed by the extension number to call directly this number.

Example: user can type *881000 to call the mobile number associated with extension 1000.

Table 34: SIP Extension Configuration ParametersMedia

SIP Settings

NAT

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.

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Can Direct Media

DTMF Mode

TEL URI

Alert-Info

Enable T.38 UDPTL

SRTP

Fax Mode

Fax To Email

Strategy

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.

If the phone has an assigned PSTN telephone number, this field should be set to “User=Phone”. “User=Phone” parameter will be attached to the

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.

Configure the Alert-Info, when UCM6100 receives an INVITE request, the

Alert-Info header field specifies an alternative ring tone to the UAS.

Enable or disable T.38 UDPTL support.

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

Select Fax mode. The default setting is “None”.

None: Disable Fax.

Fax Detect: Fax signal from the user/trunk during the call can be detected and the received Fax will be sent to the Email address configured for this extension. If no Email address can be found for the user, the Fax will be sent to the default Email address configured in

Fax setting page under Web GUICall FeaturesFax/T.38.

YesAllow Fax to Email for this extension. Faxes will be sent to the user’s email address configured in the extension’s Basic

Settings.

NoDo not send any faxes to the user’s email address configured in the extensi on’s Basic Settings.

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.

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Codec Preference

A Specific IP Address

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

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,

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

Table 35: SIP Extension Configuration ParametersFeatures

Call Transfer

Presence Status

Call Forward Unconditional

CFU Time Condition

Call Forward No Answer

Select which presence status to set for the extension and configure call forward conditions for each status. Six possible options are possible:

Available”, “Away”, “Chat”, “Custom”, “DND” and “Unavailable”.

More details at [PRESENCE].

Enable and configure the Call Forward Unconditional target number.

Available options for target number are:

• “None”: Call forward deactivated.

• “Extension”: Select an extension from dropdown list as CFU target.

• “Custom Number”: Enter a customer number as target. For example: *97.

• “Voicemail”: Select an extension from dropdown list. Incoming calls will be forwarded to voicemail of selected extension.

• “Ring Group”: Select a ring group from dropdown list as CFU target.

• “Queues”: Select a queue from dropdown list as CFU target.

• “Voicemail Group”: Select a voicemail group from dropdown list as

CFU target.

The default setting is “None”.

Select time condition for Call Forward Unconditional. CFU takes effect only during the selected time condition. The available time condition is “Office

Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or Holiday” and “Specific”.

Notes:

• “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 System

SettingsTime SettingsOffice Time/Holiday page.

Configure the Call Forward No Answer target number. Available options for target number are:

• “None”: Call forward deactivated.

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CFN Time Condition

Call Forward Busy

CFB Time Condition

• “Extension”: Select an extension from dropdown list as CFN target.

• “Custom Number”: Enter a customer number as target. For example: *97.

• “Voicemail”: Select an extension from dropdown list. Incoming calls will be forwarded to voicemail of selected extension.

• “Ring Group”: Select a ring group from dropdown list as CFN target.

• “Queues”: Select a queue from dropdown list as CFN target.

• “Voicemail Group”: Select a voicemail group from dropdown list as

CFN target.

The default setting is “None”.

Select time condition for Call Forward No Answer. The available time conditions are “Office Time”, “Out of Office Time”, “Holiday”, “Out of

Holiday”, “Out of Office Time or Holiday” and “Specific”.

Notes:

• “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 System

SettingsTime SettingsOffice Time/Holiday page.

Configure the Call Forward Busy target number. Available options for target number are:

• “None”: Call forward deactivated.

• “Extension”: Select an extension from dropdown list as CFB target.

• “Custom Number”: Enter a customer number as target. For example: *97.

• “Voicemail”: Select an extension from dropdown list. Incoming calls will be forwarded to voicemail of selected extension.

• “Ring Group”: Select a ring group from dropdown list as CFB target.

• “Queues”: Select a queue from dropdown list as CFB target.

• “Voicemail Group”: Select a voicemail group from dropdown list as

CFB target.

The default setting is “None”.

Select time condition for Call Forward Busy. The available time conditions are “Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of

Office Time or Holiday” and “Specific”.

Notes:

• “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

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Do Not Disturb

DND Time Condition

DND Whitelist

FWD Whitelist

CC Settings

Enable CC

CC Mode

CC Max Agents

CC Max Monitors

Ring Simultaneously

configuration dialog. Scroll down the add Time Condition.

• Office Time and Holiday could be configured on page System

SettingsTime SettingsOffice Time/Holiday page.

If enabled the extension will ignore all incoming calls

Select time condition for Do Not Disturb. The available time conditions are

“Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of

Office Time or Holiday” and “Specific”.

Notes:

• “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 System

SettingsTime SettingsOffice Time/Holiday page.

If DND is enabled, all calls to this extension will be rejected except the numbers listed on this list.

Note: The maximum number on the whitelist is 10.

If call forward is enabled, all calls to this extension will be forwarded except the calls coming from the specified numbers on this list.

Note: The Maximum number on the whitelist is 10.

If enabled, UCM6100 will automatically alert this extension when a called party is available, given that a previous call to that party failed for some reason. By default, it’s disabled.

Two modes for Call Completion are supported:

Normal: This extension is used as ordinary extension.

For Trunk: This extension is registered from a PBX.

Note: The default setting is “Normal”.

Configure the maximum number of CCSS agents which may be allocated for this channel. In other words, this number serves as the maximum number of CC requests this channel is allowed to make.

Note: The minimum value is 1.

Configure the maximum number of monitor structures which may be created for this device. In other words, this number tells how many callers may request CC services for a specific device at one time.

Note: The minimum value is 1.

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Ring Simultaneously

Enable this option to have an external number ring simultaneously along with the extension. If a register trunk is used for outbound, the register number will be used to be displayed for the external number as caller ID.

External Number

Time Condition for Ring

Simultaneously

Use callee DOD on FWD or

Ring Simultaneously

Monitor privilege control

Set the external number to be rang simultaneously. ‘-’ is the connection character which will be ignored.

Ring the external number simultaneously along with the extension on the basis of this time condition.

Use the DOD number when calls are being diverted/forwarded to external destinations or when ring simultaneous is configured.

Allowed to call-barging

Add members from “Available Extensions” to “Selected Extensions” and the selected extensions can spy on the used extension using feature code.

Seamless transfer privilege control

Allowed to seamless transfer

Any extensions on the UCM can perform seamless transfer. When using

Pickup In-call feature, only extensions available on the “Selected

Extensions” list can perform seamless transfer to the edited extension.

Other Settings

Ring Timeout

Auto Record

Skip Trunk Auth

Time Condition for Skip

Trunk Auth

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PBX SettingsGeneral Settings 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CDRRecording Files.

• If set to “yes”, users can skip entering the password when making outbound calls.

• If set to “By Time”, users can skip entering the password when making outbound calls during the selected time condition.

• If set to “No”, users will be asked to enter the password when making outbound calls.

If ‘Skip Trunk Auth’ is set to ‘By Time’, select a time condition during which users can skip entering password when making outbound calls.

Dial Trunk Password

Configure personal password when making outbound calls via trunk.

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Support Hot-Desking Mode

Enable LDAP

Enable WebRTC Support

Music On Hold

Enable Seamless Transfer

Permission

Call Duration Limit

Custom Call-info for Auto

Answer

If enabled, SIP Password will accept only alphabet characters and digits.

Auth ID will be changed to the same as Extension.

If enabled, the extension will be added to LDAP Phonebook PBX list.

Enable registration and call from WebRTC.

Specify which Music On Hold class to suggest to the bridged channel when putting them on hold.

Enable the seamless transfer for this extension.

Set the permission for this extension when using the seamless transfer

The maximum duration of call-blocking.

If enabled, when a call is sent to this extension from UCM, the SIP INVITE message will contain a Call-info header indicating auto answer.

Table 36: SIP Extension Configuration ParametersSpecific Time

Specific Time

Time Condition

Click to add Time Condition to configure specific time for this extension.

Create New IAX Extension

The UCM6100 supports Inter-Asterisk eXchange (IAX) protocol. IAX is used for transporting VoIP telephony sessions between servers and terminal devices. IAX is similar to SIP but also has its own characteristic.

For more information, please refer to RFC 5465.

To manually create new IAX user, go to Web GUIExtension/TrunkExtensions. Click on "Add" and a new dialog window will show for users which need to make sure first to select the extension type to be IAX

Extension before proceeding to fill in the extension information. The configuration parameters are as follows.

Table 37: IAX Extension Configuration ParametersBasic Settings

General

Extension

CallerID Number

Permission

SIP/IAX Password

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

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Enable Voicemail

Voicemail Password

Skip Voicemail Password

Verification

Send Voicemail to Email

Keep Voicemail after

Emailing

Disable This Extension

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.

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.

Allows users to send voicemail recording as .wav file attachment to specified email addresses, providing more customizable user experience.

Note: When set to “Default”, the global settings in Call Features

VoicemailVoicemail Email Settings will be used.

Whether to keep the local voicemail recording after sending them. If set to

“Default”, the global settings will be used.

Note: When set to “Default”, the global settings in Call Features

VoicemailVoicemail Email Settings will be used.

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.

User Settings

First Name

Last Name

Email Address

User Password

Language

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

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

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

Configure the password for user portal access. A random numeric password is automatically generated. It is recommended to use the randomly generated password for security purpose.

Select the voice prompt language to be used for this extension. The default setting is "Default" which is the selected voice prompt language under web

GUIPBX SettingsVoice PromptLanguage Settings. 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

SettingsVoice PromptLanguage Settings.

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Table 38: IAX Extension Configuration ParametersMedia

IAX Settings

Max Number of Calls

Require Call Token

SRTP

Fax Mode

Strategy

Codec Preference

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

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.

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

Select Fax Mode. The default setting is “None”.

None: Disable Fax. This is the default setting.

Fax Detect: Fax signal from the user/trunk during the call can be detected and the received Fax will be sent to the Email address configured for this extension. If no Email address can be found for the user, the Fax will be sent to the default Email address configured in

Fax setting page under Web GUICall FeaturesFax/T.38.

This option controls how the extension can be used on devices within different types of network.

Allow All

Device in any network can register this extension.

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.

Table 39: IAX Extension Configuration ParametersFeatures

Call Transfer

Call Forward Unconditional

CFU Time Condition

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 is “Office

Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or Holiday” and “Specific”.

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Call Forward No Answer

CFN Time Condition

Call Forward Busy

CFB Time Condition

Notes:

• “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 System

SettingsTime SettingsOffice Time/Holiday page.

Configure the Call Forward No Answer target number. If not configured, the Call Forward No Answer feature is deactivated. Default is deactivated.

Select time condition for Call Forward No Answer. The available time conditions are “Office Time”, “Out of Office Time”, “Holiday”, “Out of

Holiday”, “Out of Office Time or Holiday” and “Specific”.

Notes:

• “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 System

SettingsTime SettingsOffice 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 conditions are “Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of

Office Time or Holiday” and “Specific”.

Notes:

• “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 System

SettingsTime SettingsOffice Time/Holiday page.

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Ring Simultaneously

Ring Simultaneously

External Number

Time Condition for Ring

Simultaneously

Other Settings

Enable this option to have an external number ring simultaneously along with the extension. If a register trunk is used for outbound, the register number will be used to be displayed for the external number as caller ID number.

Set the external number to be rang simultaneously. ‘-’ is the connection character which will be ignored.

Ring the external number simultaneously along with the extension on the basis of this time condition.

Ring Timeout

Auto Record

Skip Trunk Auth

Time Condition for Skip

Trunk Auth

Dial Trunk Password

Enable LDAP

Music On Hold

Call Duration Limit

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PBX SettingsGeneral Settings: 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CDRRecording Files.

• If set to “Yes”, users can skip entering the password when making outbound calls.

• If set to “By Time”, users can skip entering the password when making outbound calls during the selected time condition.

• If set to “No”, users will be asked to enter the password when making outbound calls.

If “Skip Trunk Auth” is set to “By Time”, select a time condition during which users can skip entering password when making outbound calls.

Configure personal password when making outbound calls via trunk.

If enabled, the extension will be added to LDAP Phonebook PBX lists.

Configure the Music On Hold class to suggest to the bridged channel when putting them on hold.

The maximum duration of call-blocking.

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Table 40: IAX Extension Configuration ParametersSpecific Time

Specific Time

Time Condition Click to add Time Condition to configure specific time for this extension.

Create New FXS Extension

The UCM6100 supports Foreign eXchange Subscriber (FXS) interface. FXS is used when user needs to connect analog phone lines or FAX machines to the UCM6100.

To manually create new FXS user, go to Web GUIExtension/TrunkExtensions. Click on "Add" and a new dialog window will show for users which need to make sure first to select the extension type to be FXS

Extension before proceeding to fill in the extension information. The configuration parameters are as follows.

Table 41: FXS Extension Configuration ParametersBasic Settings

General

Extension

Analog Station

CallerID Number

Permission

Enable Voicemail

The extension number associated with the user.

Select the FXS port to be assigned for this extension.

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".

Voicemail Password

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.

Skip Voicemail Password

Verification

Disable This Extension

When user dials voicemail code, the password verification IVR is skipped.

If enabled, this would allow one-button voicemail access. By default, this option is disabled.

If selected, this extension will be disabled on the UCM6100.

Note:

The disabled extension still exists on the PBX but can’t be used on the end device.

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User Settings

First Name

Last Name

Email Address

User Password

Language

Analog Settings

Call Waiting

User ‘#’ as SEND

RX Gain

TX Gain

MIN RX Flash

MAX RX Flash

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

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

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

Configure the password for user portal access. A random numeric password is automatically generated. It is recommended to use the randomly generated password for security purpose.

Select the voice prompt language to be used for this extension. The default setting is "Default" which is the selected voice prompt language under web

GUIPBX SettingsVoice PromptLanguage Settings. 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

SettingsVoice PromptLanguage Settings.

Table 42: FXS Extension Configuration ParametersMedia

Configure to enable/disable call waiting feature. The default setting is "No".

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

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

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

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

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

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Enable Polarity Reversal

Echo Cancellation

3-Way Calling

Send CallerID After

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 hang up on a polarity reversal. The default 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 mean 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. Default setting is 1.

For FXS extension, there are three options available in Fax Mode. The default setting is “None”.

Fax Mode

None: Disable Fax.

Fax Detect: Fax signal from the user/trunk during the call can be detected and the received Fax will be sent to the Email address configured for this extension. If no Email address can be found for the user, the Fax will be sent to the default Email address configured in

Fax setting page under Web GUICall FeaturesFax/T.38.

Fax Gateway: If selected, the UCM6100 can support conversation and processing of Fax data from T.30 to T.38 or T.38 to T.30. This feature is only available for FXS or FXO port.

Table 43: FXS Extension Configuration ParametersFeatures

Call Transfer

Call Forward Unconditional

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

Select time condition for Call Forward Unconditional. CFU takes effect only during the selected time condition. The available time conditions are

“Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of

Office Time or Holiday” and “Specific”.

Notes:

• “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.

• Office Time and Holiday could be configured on page System

SettingsTime SettingsOffice 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 conditions are “Office Time”, “Out of Office Time”, “Holiday”, “Out of

Holiday”, “Out of Office Time or Holiday” and “Specific”.

Notes:

• “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 System

SettingsTime SettingsOffice 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 conditions are “Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of

Office Time or Holiday” and “Specific”.

Notes:

• “Specific” has higher priority to “Office Times” if there is a conflict in terms of time period.

• Specific time can be configured on the bottom of the extension configuration dialog. Scroll down the add Time Condition for specific time.

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CC Settings

Enable CC

Ring Simultaneously

Ring Simultaneously

External Number

Time Condition for Ring

Simultaneously

Hotline

Enable Hotline

Hotline Number

Hotline Type

Other Settings

Ring Timeout

Auto Record

Skip Trunk Auth

• Office Time and Holiday could be configured on page System

SettingsTime SettingsOffice Time/Holiday page.

If enabled, UCM6100 will automatically alert this extension when a called party is available, given that a previous call to that party failed.

Enable this option to have an external number ring simultaneously along with the extension.

If a register trunk is used for outbound, the register number will be used to be displayed for the external number as caller ID number.

Set the external number to be rang simultaneously. ‘-’ is the connection character which will be ignored.

Ring the external number simultaneously along with the extension on the basis of this time condition.

If enabled, hotline dialing plan will be activated, a pre-configured number will be used according to the selected Hotline Type.

Configure the Hotline Number

Configure the Hotline Type:

Immediate Hotline: When the phone is off-hook, UCM6100 will immediately dial the preset number

Delay Hotline: When the phone is off-hook, if there is no dialing within

5 seconds, UCM6100 will dial the preset number.

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PBX SettingsGeneral Settings: 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CDRRecording Files.

• If set to “Yes”, users can skip entering the password when making outbound calls.

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Time Condition for Skip

Trunk Auth

Dial Trunk Password

Enable LDAP

Music On Hold

Call Duration Limit

• If set to “By Time”, users can skip entering the password when making outbound calls during the selected time condition.

• If set to “No”, users will be asked to enter the password when making outbound calls.

If “Skip Trunk Auth” is set to “By Time”, select a time condition during which users can skip entering password when making outbound calls.

Configure personal password when making outbound calls via trunk.

If enabled, this extension will be added to LDAP Phonebook PBX list; if disabled, this extension will be skipped when creating LDAP Phonebook.

Select which Music On Hold class to suggest to extension when putting the active call on hold.

Configure the maximum duration of call-blocking.

Table 44: FXS Extension Configuration ParametersSpecific Time

Specific Time

Time Condition

Click to add Time Condition to configure specific time for this extension.

Batch Add Extensions

Batch Add SIP Extensions

In order to add multiple SIP extensions, “Batch add” can be used to create standardized SIP extension accounts. However, unique extension user name can’t be set using Batch add.

Under Web GUIExtension/TrunkExtensions, click on "Add” and select extension type as SIP extension, and “add method” as Batch.

Table 45: Batch Add SIP Extension Parameters

General

Extension

Create Number

Extension Incrementation

Permission

Enable Voicemail

Enable WebRTC Support

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.

Specify the interval between extensions as preferred when creating a batch of extension.

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".

Enable WebRTC support.

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SIP/IAX Password

Voicemail Password

CallerID Number

Ring Timeout

Auto Record

Skip Voicemail Password

Verification

Music On Hold

Enable LDAP

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 CallerID Number when adding Batch Extensions.

Use Extension as Number

Users can choose to use the extension number as CallerID by typing on this filed ‘e’.

Use as Number

Users can choose to set a specific number instead of using the extension number as CallerID.

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PBX SettingsGeneral Settings: 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CDRRecording 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.

Select which Music On Hold class to suggest to extensions when putting them on hold.

If enabled, the batch added extensions will be added to LDAP Phonebook

PBX list; if disabled, the batch added extensions will be skipped when creating LDAP Phonebook.

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Enable WebRTC Support

Call Duration Limit

SIP Settings

If enabled, extensions will be able to login to user portal and use Web RTC features.

Configure the maximum duration of call-blocking.

NAT

Can Direct Media

DTMF Mode

Enable Keep-alive

Keep-alive Frequency

TEL URI

Concurrent Registrations

Other Settings

SRTP

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 it’s "Inband".

If enabled, empty SDP packet will be sent to the SIP server periodically to keep the NAT port open. The default setting is "Yes".

Configure the number of seconds for the host to be up for Keep-alive. The default setting is 60 seconds.

If the end device/phone has an assigned PSTN telephone number, this field should be set to "User=Phone". Then a "User=Phone" parameter will be 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.

The maximum endpoints which can be registered into this extension. For security concerns, the default value is 1.

Fax Mode

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

Select Fax mode for this user. The default setting is “None”.

None: Disable Fax.

Fax Detect: Fax signal from the user/trunk during the call can be detected and the received Fax will be sent to the Email address configured for this extension. If no Email address can be found for the user, the Fax will be sent to the default Email address configured in

Fax setting page under Web GUICall FeaturesFax/T.38.

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Strategy

Enable T.38 UDPTL

Skip Trunk Auth

Codec Preference

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".

Enable or disable T.38 UDPTL Support.

If enable “All”, users don’t need to enter password when making calls. If enable “Follow Me”, they can use follow me without authentication.

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, OPUS,

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

Batch Add IAX Extensions

Under Web GUIExtension/TrunkExtensions, click on “Add”, then select extension type as IAX

Extension and the add method to be Batch.

Table 46: Batch Add IAX Extension Parameters

General

Start Extension

Create Number

Permission

Enable Voicemail

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”.

SIP/IAX Password

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.

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Voicemail Password

Ring Timeout

Auto Record

Skip Voicemail Password

Verification

Music On Hold

Enable LDAP

Call Duration Limit

IAX Settings

Max Number of Calls

Require Call Token

• 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PBX SettingsGeneral Settings: 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CDRRecording 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.

Select which Music On Hold class to suggest to extensions when putting them on hold.

If enabled, the batch added extensions will be added to LDAP Phonebook

PBX list; if disabled, the batch added extensions will be skipped when creating LDAP Phonebook.

Configure the maximum duration of call-blocking.

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

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".

Other Settings

SRTP

Fax Mode

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

Select Fax Mode for this user. The default setting is “None”.

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Strategy

Skip Trunk Auth

Codec Preference

Search and Edit Extension

None: Disable Fax.

Fax Detect: Fax signal from the user/trunk during the call can be detected and the received Fax will be sent to the Email address configured for this extension. If no Email address can be found for the user, the Fax will be sent to the default Email address configured in

Fax setting page under Web GUICall FeaturesFax/T.38.

This option controls how the extension can be used on devices within different types of network.

Allow All

Device in any network can register this extension.

Local Subnet Only

Only the user in specific subnet can register this extension. Up to three subnet addresses can be specified.

A Specific IP Address

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

The default setting is "Allow All".

If enable “All”, users do not need to enter password when making an outbound call. If enable “Follow Me”, the call can dial out via follow me without password.

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

PCMU, PCMA, GSM, AAL2-G.726-32, G.722, G.729, G.723, iLBC,

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

All the UCM6100 extensions are listed under Web GUIExtension/TrunkExtensions, with status,

Extension, CallerID Name, Terminal Type (SIP, SIP(WebRTC), 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. User can search an extension by specifying the extension number to find an extension quickly.

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Figure 87: Manage Extensions

Status

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

Green:

Blue:

Yellow:

Grey:

Edit single extension

Free

Ringing

In Use

Unavailable (the extension is not registered or disabled on the PBX)

Click on to start editing the extension parameters.

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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Value-added FeaturesZero ConfigAuto Provisioning Settings.

Delete single extension

Click on to delete the extension. Or select the checkbox of the extension and then click on "Delete

Selected Extensions".

Modify selected extensions

Select the checkbox for the extension(s). Then click on "Modify Selected Extensions" to edit the extensions in a batch.

Delete selected extensions

Select the checkbox for the extension(s). Then click on "Delete Selected Extensions" to delete the extension(s).

Export Extensions

The extensions configured on the UCM6100 can be exported to csv format file with selected technology

"SIP", "IAX" or "FXS". Click on "Export Extensions" button and select technology in the prompt below.

Figure 88: 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.

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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 89: Import Extensions

4.

Select the option in "On Duplicate Extension" to define how the duplicate extension(s) in the imported csv file should be treated by the PBX.

Skip: Duplicate extensions in the csv file will be skipped. The PBX will keep the current extension information as previously configured without change.

Delete and Recreate: The current extension previously configured will be deleted and the duplicate extension in the csv file will be loaded to the PBX.

Update Information: The current extension previously configured in the PBX will be kept. However, if the duplicate extension in the csv file has different configuration for any options, it will override the configuration for those options in the extension.

5.

Click on “choose file to upload” to select csv file from local directory in the PC.

6.

Click on "Apply Changes" to apply the imported file on the UCM6100.

Example of file to import:

Field

Extension

Technology

Enable Voicemail

Figure 90: Import File

Table 47: SIP extensions Imported File Example

Supported values

Digits

SIP/SIP(WebRTC) yes/no

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CallerID Number

SIP/IAX Password

Voicemail Password

Skip Voicemail Password Verification

Ring Timeout

SRTP

Fax Mode

Strategy

Local Subnet 1

Local Subnet 2

Local Subnet 3

Local Subnet 4

Local Subnet 5

Local Subnet 6

Local Subnet 7

Local Subnet 8

Local Subnet 9

Local Subnet 10

Specific IP Address

Skip Trunk Auth

Codec Preference

Permission

NAT

DTMF Mode

Insecure

Enable Keep-alive

Keep-alive Frequency

AuthID

TEL URI

Call Forward Busy

Call Forward No Answer

Call Forward Unconditional

Support Hot-Desking Mode

Dial Trunk Password

Disable This Extension

CFU Time Condition

Digits

Alphanumeric characters

Digits yes/no

Empty/ 3 to 600 (in second) yes/no

None/Fax Detection

Allow All/Local Subnet Only/A Specific IP Address

IP address/Mask

IP address/Mask

IP address/Mask

IP address/Mask

IP address/Mask

IP address/Mask

IP address/Mask

IP address/Mask

IP address/Mask

IP address/Mask

IP address yes/no/bytime

PCMU,PCMA,GSM,G.726,G.722,G.729,H.264,ILBC,AAL2-

G.726-32,ADPCM,G.723,H.263,H.263p,vp8,opus

Internal/Local/National/International yes/no

RFC2833/info/inband/auto

Port

Yes/no

Value from 1-3600

Alphanumeric value without special characters

Disabled/user=phone/enabled

Digits

Digits

Digits

Yes/no

Digits

Yes/no

All time/Office time/out of office time/holiday/out of

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CFN Time Condition

CFB Time Condition

holiday/out of office time or holiday/specific time

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

Default/ringbacktone_default

Music On Hold

CC Agent Policy

CC Monitor Policy

CCBS Available Timer

CCNR Available Timer

CC Offer Timer

CC Max Agents

CC Max Monitors

If CC is disabled use: never

If CC is set to normal use: generic

If CC is set to trunk use: native

Generic/never

3600/4800

3600/7200

60/120

Value from 1-999

Value from 1-999

Ring simultaneously

External Number

Yes/no

Digits

Time Condition for Ring Simultaneously

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

Time Condition for Skip Trunk Auth

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

Enable LDAP

Enable T.38 UDPTL

Yes/no

Yes/no

Max Contacts

Enable WebRTC

Alert-Info

Do Not Disturb

DND Time Condition

Values from 1-10

Yes/no

None/Ring 1/Ring2/Ring3/Ring 4/Ring 5/Ring 6/Ring 7/

Ring 8/Ring 9/Ring 10/bellcore-dr1/bellcore-dr2/ bellcore-dr3/ bellcore-dr4/ bellcore-dr5/custom

Yes/no

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

Custom Auto answer

Do Not Disturb Whitelist

User Password

First Name

Last Name

Email Address

Language

Yes/no

Empty/digits

Alphanumeric characters.

Alphanumeric without special characters.

Alphanumeric without special characters.

Email address

Default/en/zh

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Phone Number

Call-Barging Monitor

Seamless Transfer Members

Digits

Extensions allowed to call barging

Extensions allowed to seamless transfer

Table 48: IAX extensions Imported File Example

Field

Extension

Technology

Enable Voicemail

CallerID Number

SIP/IAX Password

Voicemail Password

Skip Voicemail Password Verification

Ring Timeout

SRTP

Fax Mode

Strategy

Local Subnet 1

Local Subnet 2

Local Subnet 3

Local Subnet 4

Local Subnet 5

Local Subnet 6

Local Subnet 7

Local Subnet 8

Local Subnet 9

Local Subnet 10

Specific IP Address

Skip Trunk Auth

Codec Preference

Permission

NAT

Call Forward Busy

Call Forward No Answer

Call Forward Unconditional

Require Call Token

Supported values

Digits

IAX yes/no

Digits

Alphanumeric characters

Digits yes/no

Empty/ 3 to 600 (in second) yes/no

None/Fax Detection

Allow All/Local Subnet Only/A Specific IP Address

IP address/Mask

IP address/Mask

IP address/Mask

IP address/Mask

IP address/Mask

IP address/Mask

IP address/Mask

IP address/Mask

IP address/Mask

IP address/Mask

IP address yes/no/bytime

PCMU,PCMA,GSM,G.726,G.722,G.729,H.264,ILBC,AAL2-

G.726-32,ADPCM,G.723,H.263,H.263p,vp8,opus

Internal/Local/National/International yes/no

Digits

Digits

Digits

Yes/no/auto

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Max Number of Calls

Dial Trunk Password

Disable This Extension

CFU Time Condition

Values from 1-512

Digits

Yes/no

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

CFN Time Condition

CFB Time Condition

Limit Max time (s)

Do Not Disturb

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

Default/ringbacktone_default

Music On Hold

Ring simultaneously

External Number

Time Condition for Ring Simultaneously

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

Time Condition for Skip Trunk Auth

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

Enable LDAP

Yes/no

Digits

Yes/no empty

DND Time Condition

Yes/no

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

Do Not Disturb Whitelist

User Password

First Name

Last Name

Email Address

Language

Phone Number

Call-Barging Monitor

Seamless Transfer Members

Empty/digits

Alphanumeric characters.

Alphanumeric without special characters.

Alphanumeric without special characters.

Email address

Default/en/zh

Digits

Extensions allowed to call barging

Extensions allowed to seamless transfer

Field

Extension

Technology

Analog Station

Enable Voicemail

CallerID Number

Table 49: FXS extensions Imported File Example

Supported values

Digits

FXS

FXS1/FXS2 yes/no

Digits

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Voicemail Password

Skip Voicemail Password Verification

Ring Timeout

Auto Record

Fax Mode

Skip Trunk Auth

Permission

Call Forward Busy

Call Forward No Answer

Call Forward Unconditional

Call Waiting

Use # as SEND

RX Gain

TX Gain

MIN RX Flash

MAX RX Flash

Enable Polarity Reversal

Echo Cancellation

3-Way Calling

Send CallerID After

Dial Trunk Password

Disable This Extension

CFU Time Condition

CFN Time Condition

CFB Time Condition

Music On Hold

CC Agent Policy

CC Monitor Policy

CCBS Available Timer

CCNR Available Timer

CC Offer Timer

CC Max Agents

Digits yes/no

Empty/ 3 to 600 (in second) yes/no

None/Fax Detection

Yes/no/bytime

Internal/Local/National/International

Digits

Digits

Digits

Yes/no

Yes/no

Values from -306

Values from -306

Values from: 30 – 1000

Values from: 40 – 2000

Yes/no

On/Off/32/64/128/256/512/1024

Yes/no

1/2 digits

Yes/no

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

Default/ringbacktone_default

If CC is disabled use: never

If CC is set to normal use: generic

If CC is set to trunk use: native

Generic/never

3600/4800

3600/7200

60/120

Value from 1-999

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CC Max Monitors

Ring simultaneously

External Number

Value from 1-999

Yes/no

Digits

Time Condition for Ring Simultaneously

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

Time Condition for Skip Trunk Auth

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

Enable LDAP

Enable Hotline

Hotline Type

Hotline Number

Yes/no

Yes/no

Immediate hotline/delay hotline digits

Limit Max time (s)

Do Not Disturb

DND Time Condition

empty

Yes/no

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

Empty/digits

Alphanumeric characters.

Do Not Disturb Whitelist

User Password

First Name

Last Name

Email Address

Language

Alphanumeric without special characters.

Alphanumeric without special characters.

Email address

Default/en/zh

Phone Number

Call-Barging Monitor

Seamless Transfer Members

Digits

Extensions allowed to call barging

Extensions allowed to seamless transfer

The CSV file should contain all the above fields, if one of them is missing or empty, the UCM6100 will display the following error message for missing fields.

Figure 91: Import Error

E-mail Notification

Once the extensions are created with Email address, the PBX administrator can click on button “E-mail

Notification” to send the account registration and configuration information to the user. Please make sure

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Email setting under Web GUISystem SettingsEmail Settings is properly configured and tested on the UCM6100 before using “E-mail Notification”.

When click on “E-mail Notification” button, the following message will be prompted in the web page. Click on OK to confirm sending the account information to all users’ Email addresses.

Figure 92: E-mail Notification - Prompt Information

The user will receive Email including account registration information and LDAP configuration. A QR code is also generated for Mobile applications to scan it and get automatically provisioned. QR code provisioning is supported on Grandstream Softphone GS Wave Android

TM

application and iPhone application.

Figure 93: Account Registration Information and QR Code

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Figure 94: LDAP Client Information and QR Code

Multiple Registrations per Extension

UCM6100 supports multiple registrations per extension so that users can use the same extension on devices in different locations.

Figure 95: Multiple Registrations per Extension

This feature can be enabled by configuring option “Concurrent Registrations” under Web

GUIExtension/TrunkEdit Extension. The default value is set to 1 for security purpose. Maximum is

10.

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Figure 96: Extension - Concurrent Registration

SMS Message Support

The UCM6100 provides built-in SIP SMS message support. For SIP end devices such as Grandstream

GXP or GXV phones that supports SIP message, after an UCM6100 account is registered on the end device, the user can send and receive SMS message. Please refer to the end device documentation on how to send and receive SMS message.

SMS Message support is a new feature added since firmware 1.0.10.x which is built with Asterisk 13.

Figure 97: SMS Message Support

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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Extension/TrunkExtension Groups.

• Click on

to create a new extension group.

• Click on

to edit the extension group.

• Click on

to delete the extension group.

Select extensions from the list on the left side to the right side.

Figure 98: Edit Extension Group

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Using Extension Groups

Here is an example where the extension group can be used. Go to Web

GUIExtension/TrunkOutbound Routes and select "Enable Filter on Source Caller ID". Both single extensions and extension groups will show up for users to select. This way we can restrict the permission of a specific outbound rule to a certain group of people. As Ex: only Sales agents are allowed to make international calls if the rule has the matching number pattern for international dialing.

Figure 99: Select Extension Group in Outbound Route

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ANALOG TRUNKS

Go to Web GUIExtension/TrunkAnalog 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

FXO Port

Trunk Name

SLA Mode

Barge Allowed

Hold Access

to delete the analog trunk.

Analog Trunk Configuration

The analog trunk options are listed in the table below.

Table 50: Analog Trunk Configuration Parameters

Select the FXO ports 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 hold 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

Polarity on Answer Delay

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 “hang up” on a polarity reversal. Default 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.

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Current Disconnect

Threshold (ms)

Ring Timeout

RX Gain

TX Gain

Use CallerID

Caller ID Scheme

FXO Dial Delay(ms)

Auto Record

Disable This Trunk

DAHDI Out Line Selection

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”.

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CDRRecording Files.

If selected, the trunk will be disabled.

This is to implement analog trunk outbound line selection strategy. Three options are available:

Ascend

When the call goes out from this analog trunk, it will always try to use the first idle FXO port. The port order that the call will use to go out would be port 1port 2port 10port 16. Every time it will start with port 1 (if it's idle).

Poll

When the call goes out from this analog trunk, it will use the port that is not used last time. And it will always use the port in the order of port

121016121016121016..., following the last port being used.

Descend

When the call goes out from this analog trunk, it will always try to use the last idle FXO port. The port order that the call will use to go out would be port 16port 10port 2port 1. Every time it will start with port 16 (if it is idle).

The default setting is “Ascend” mode.

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Tone Settings

Busy Detection

Busy Tone Count

Congestion Detection

Congestion Count

Tone Country

Busy Tone

Congestion Tone

PSTN Detection

Busy Detection is used to detect far end hang up or for detecting busy signal. The default setting is "Yes".

If "Busy Detection" is enabled, users can specify the number of busy tones to be played before hanging up. The default setting is 2. Better results might be achieved if set to 4, 6 or even 8. Please note that the higher the number is, the more time is needed to hang up the channel. However, this might lower the probability to get random hang up.

Congestion detection 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Extension/TrunkAnalog 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 100: UCM6100 FXO Tone Settings

4.

Click on "Detect" to start PSTN detection.

Figure 101: UCM6100 PSTN Detection

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• 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 102: UCM6100 PSTN Detection: Auto Detect

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

Figure 103: UCM6100 PSTN Detection: Semi-Auto Detect

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Detect Model: Semi-auto Detect.

Source Channel: The source channel to be detected.

Destination Number: The number to be dialed for detecting. This number could be a cell phone number or other PSTN number that can be reached from the source channel PSTN number.

5.

Click "Detect" to start detecting. The source channel will initiate a call to the destination number. For

"Auto Detect", the call will be automatically answered. For "Semi-auto Detect", the UCM6100 web GUI will display prompt to notify the user to answer or hang up the call to finish the detecting process.

6.

Once done, the detected result will show. Users could save the detecting result as the current UCM6100 settings.

Detect Model

Source Channel

Destination Channel

Destination Number

Table 51: PSTN Detection for Analog Trunk

Select "Auto Detect" or "Semi-auto Detect" for PSTN detection.

Auto Detect

Please make sure two or more channels are connected to the

UCM6100 and in idle status before starting the detection. During the detection, one channel will be used as caller (Source Channel) and another channel will be used as callee (Destination Channel). The

UCM6100 will control the call to be established and hang up between caller and callee to finish the detection.

Semi-auto Detect

Semi-auto detection requires answering or hanging up the call manually. Please make sure one channel is connected to the

UCM6100 and in idle status before starting the detection. During the detection, source channel will be used as caller and send the call to the configured Destination Number. Users will then need follow the prompts in web GUI to help finish the detection.

The default setting is "Auto Detect".

Select the channel to be detected.

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

Configure the number to be called to help the detection.

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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.

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VOIP TRUNKS

VoIP trunks can be configured in UCM6100 under Web GUIExtension/TrunkVoIP Trunks. Once created, the VoIP trunks will be listed with Provider Name, Type, Hostname/IP, Username and Options to edit/detect the trunk.

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

• Click on

to configure detailed parameters for the VoIP trunk.

• Click on

• Click on

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

to start LDAP Sync.

• Click on

to delete the VoIP trunk.

For VoIP trunk example, please refer to the document in the following link: http://www.grandstream.com/sites/default/files/Resources/ucm_to_ucm_peer_guide.pdf

The VoIP trunk options are listed in the table below.

Type

Provider Name

Host Name

Keep Original CID

Keep Trunk CID

NAT

Disable This Trunk

Table 52: Create New SIP Trunk

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.

Keep the CID from the inbound call when dialing out. This setting will override “Keep Trunk CID” option. Please make sure that the peer PBX at the other side supports to match user entry using “username” field from authentication line.

If enabled, the trunk CID will not be overridden by extension’s CID when the extension has CID configured. The default setting is “No”.

Turn on this setting when the PBX is using public IP and communicating with devices behind NAT. If there is one-way audio issue, usually it is related to NAT configuration or SIP/RTP port support on the firewall.

If checked, the trunk will be disabled.

Note: If a current SIP trunk is disabled, UCM will send UNREGISTER message (REGISTER message with expires=0) to the SIP provider.

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TEL URI

Caller ID

Need Registration

Username

Password

Auth ID

Auto Record

Basic Settings

Provider Name

Host Name

Transport

Keep Original CID

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 is 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 SettingsGeneral Settings will be used.

Select whether the trunk needs to register on the external server or not when "Register SIP Trunk" type is selected. The default setting is No.

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

"Register SIP Trunk" type is selected.

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

"Register SIP Trunk" is selected.

Enter the Authentication ID for "Register SIP Trunk" type.

Enable automatic recording for the calls using this trunk (for SIP trunk only). The default setting is disabled. The recording files can be accessed under web GUICDRRecording Files.

Table 53: SIP Register Trunk Configuration Parameters

Configure a unique label to identify this trunk when listed in outbound rules, inbound rules and etc.

Configure the IP address or URL for the VoIP provider’s server of the trunk.

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

• UDP

• TCP

• TLS

Keep the CID from the inbound call when dialing out. This setting will override “Keep Trunk CID” option. Please make sure that the peer PBX at the other side supports to match user entry using “username” field from authentication line.

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Keep Trunk CID

NAT

Disable This Trunk

TEL URI

Need Registration

Allow outgoing calls if registration failure

CallerID Name

From Domain

From User

Username

Password

Auth ID

Auth Trunk

Auto Record

If enabled, the trunk CID will not be overridden by extension’s CID when the extension has CID configured. The default setting is “No”.

Turn on this option when the PBX is using public IP and communicating with devices behind NAT. If there is one-way audio issue, usually it’s related to NAT configuration or SIP/RTP port configuration on the firewall.

If selected, the trunk will be disabled.

Note: If a current SIP trunk is disabled, UCM will send UNREGISTER message (REGISTER message with expires=0) to the SIP provider.

If the trunk has an assigned PSTN telephone number, this field should be set to "User=Phone". Then a "User=Phone" parameter will be attached to 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.

If enabled outgoing calls even if the registration to this trunk fail will still be able to go through.

Note that if we uncheck “Need Registration” option, this option will be ignored.

Configure the new name of the caller when the extension has no CallerID

Name configured.

Configure the actual domain name where the extension comes from. This can be used to override the “From” Header.

For example, "trunk.UCM6100.provider.com" is the From Domain in From

Header: sip:[email protected]

Configure the actual user name of the extension. This can be used to override the “From” Header. There are cases where there is a single ID for registration (single trunk) with multiple DIDs.

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

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.

If enabled, the UCM will send 401 response to the incoming call to authenticate the trunk.

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CDRRecording Files.

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Advanced Settings

Codec Preference

Send PPI Header

PPI Mode

Send PAI Header

PAI Header

Outbound Proxy Support

Outbound Proxy

Remove OBP From Route

DID Mode

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,

OPUS, ADPCM, H.264, H.263, H.263p.

If enabled, the SIP INVITE message sent to the trunk will contain PPI (P-

Preferred-Identity) header. The default setting is “No”.

Note:

“Send PPI Header” and “Send PAI Header” cannot be enabled at the same time. Only one of the two headers are allowed to be contained in the SIP

INVITE message.

Default

– Include the trunk’s preferred CID (configured in Basic

Settings) in the PPI Header

Original CID

– Include the original CID in the PPI Header

DOD Number

– Include the trunk’s DOD number in the PPI

Header. If no DOD number has been set, the trunk’s preferred

CID will be used.

If enabled, the SIP INVITE message sent to the trunk will contain PAI (P-

Asserted-Identity) header including configured PAI Header. The default setting is “No”.

Note:

“Send PPI Header” and “Send PAI Header” cannot be enabled at the same time. Only one of the two headers are allowed to be contained in the SIP

INVITE message.

If “Send PAI Header” is enabled and “PAI Header” is configured as

“123456” for instance, the PAI header in the SIP message sent from the

UCM will contain “123456”. If “Send PAI Header” is enabled and “PAI

Header” is configured as “empty”, the PAI header in the SIP message sent from the UCM will contain the original CID.

Note:

“Send PAI Header” needs to be enabled in order to use this feature

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.

If enabled, “Route:” header will be removed from SIP requests.

By default, this option is disabled.

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".

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DTMF Mode

Enable Heartbeat Detection

Heartbeat Frequency

Maximum Number of Call

Lines

Fax Mode

SRTP

CC Settings

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 GUIPBX

SettingsSIP SettingsToS.

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 Heartbeat Detection" 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 setting is 0, which means no limit.

Select Fax mode. The default setting is “None”.

None: Disable Fax.

Fax Detect: Fax signal from the user/trunk during the call can be detected and the received Fax will be sent to the Email address configured for this extension. If no Email address can be found for the user, the Fax will be sent to the default Email address configured in

Fax setting page under Web GUICall FeaturesFax/T.38.

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

Enable CC

CC Max Agents

CC Max Monitors

If enabled, the system will automatically alert the user when a called party is available, given that a previous call to that party failed for some reason.

Configure the maximum number of CCSS agents which may be allocated for this channel. In other words, this number serves as the maximum number of CC requests this channel is allowed to make. The minimum value is 1.

Configure the maximum number of monitor structures which may be created for this device. In other words, this number tells how many callers may request CC services for a specific device at one time. The minimum value is 1.

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Basic Settings

Provider Name

Host Name

Transport

Keep Original CID

Keep Trunk CID

NAT

Disable This Trunk

TEL URI

Caller ID

Table 54: SIP Peer Trunk Configuration Parameters

Configure a unique label to identify this trunk when listed in outbound rules, inbound rules and etc.

Configure the IP address or URL for the VoIP provider’s server of the trunk.

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

• UDP

• TCP

• TLS

Keep the CID from the inbound call when dialing out, this setting will override “Keep Trunk CID” option. Please make sure that the peer PBX at the other side supports to match user entry using “username” field from authentication line.

If enabled, the trunk CID will not be overridden by extension’s CID when the extension has CID configured. The default setting is “No”.

Turn on this option when the PBX is using public IP and communicating with devices behind NAT. If there is one-way audio issue, usually it’s related to NAT configuration or SIP/RTP port configuration on the firewall.

If selected, the trunk will be disabled.

Note:

If a current SIP trunk is disabled, UCM will send UNREGISTER message

(REGISTER message with expires=0) to the SIP provider.

If the trunk has an assigned PSTN telephone number, this field should be set to "User=Phone". Then a "User=Phone" parameter will be attached to 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 SettingsGeneral Settings will be used.

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CallerID Name

Auto Record

Configure the name of the caller to be displayed when the extension has no CallerID Name configured.

Enable automatic recording for the calls using this trunk (for SIP trunk only). The default setting is disabled. The recording files can be accessed under web GUICDRRecording Files.

Advanced Settings

Codec Preference

DID Mode

DTMF Mode

Enable Heartbeat Detection

Heartbeat Frequency

Maximum Number of Call

Lines

Fax Mode

SRTP

Sync LDAP Enable

Select audio and video codec for the VoIP trunk. The available codecs are:

PCMU, PCMA, GSM, AAL2-G.726-32, G.726, G.722, G.729, G.723, iLBC,

ADPCM, H.264, H.263, H.263p.

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

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

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 GUIPBX

SettingsSIP SettingsToS.

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 Heartbeat Detection" 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.

Select Fax mode. The default setting is “None”.

None: Disable Fax.

Fax Detect: Fax signal from the user/trunk during the call can be detected and the received Fax will be sent to the Email address configured for this extension. If no Email address can be found for the user, the Fax will be sent to the default Email address configured in

Fax setting page under Web GUICall FeaturesFax/T.38.

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".

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Sync LDAP Password

Sync LDAP Port

LDAP Outbound Rule

LDAP Dialed Prefix

CC Settings

Enable CC

CC Max Agents

CC Max Monitors

Type

Provider Name

Host Name

Keep Trunk CID

Username

Password

Disable This Trunk

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 to ensure successful synchronization.

Configure the port used LDAP sync feature between two peer UCM6100.

Specify an outbound rule for LDAP sync feature. The UCM6100 will modify the remote contacts by adding prefix parsed from this rule.

Specify the prefix for LDAP sync feature. The UCM6100 will automatically modify the remote contacts by adding this prefix.

If enabled, the system will automatically alert the user when a called party is available, given that a previous call to that party failed for some reason.

Configure the maximum number of CCSS agents which may be allocated for this channel. In other words, this number serves as the maximum number of CC requests this channel is allowed to make. The minimum value is 1.

Configure the maximum number of monitor structures which may be created for this device. In other words, this number tells how many callers may request CC services for a specific device at one time. The minimum value is 1.

Table 55: Create New IAX Trunk

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.

Table 56: IAX Register Trunk Configuration Parameters

Basic Settings

Provider Name

Host Name

Configure a unique label to identify this trunk when listed in call rules.

Configure the IP address or URL for the VoIP provider’s server of the trunk.

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Keep Trunk CID

Disable This Trunk

Caller ID

CallerID Name

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:

• 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 SettingsGeneral Settings will be used.

Configure the name of the caller to be displayed when the extension has no CallerID Name configured.

Username

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

Password

Advanced Settings

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

Codec Preference

Enable Heartbeat Detection

Heartbeat Frequency

Maximum Number of Call

Lines

Fax Mode

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 Heartbeat Detection" 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 setting is 0, which means no limited.

Select Fax mode. The default setting is “None”.

None: Disable Fax.

Fax Detect: Fax signal from the user/trunk during the call can be detected and the received Fax will be sent to the Email address configured for this extension. If no Email address can be found for the user, the Fax will be sent to the default Email address configured in

Fax setting page under Web GUICall FeaturesFax/T.38.

Table 57: IAX Peer Trunk Configuration Parameters

Basic Settings

Provider Name

Configure a unique label to identify this trunk when listed in outbound rules, inbound rules and etc.

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Host Name

Keep Trunk CID

Disable This Trunk

Caller ID

CallerID Name

Configure the IP address or URL for the VoIP provider’s server of the trunk.

If enabled, the trunk CID will not be overridden by extension's CID when the extension has CID configured. The default setting is "No".

If selected, the trunk will be disabled.

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

When making outgoing calls, the following rules are used to determine 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 SettingsGeneral Settings will be used.

Configure the name of the caller to be displayed when the extension has no CallerID Name configured.

Advanced Settings

Codec Preference

Enable Heartbeat Detection

Heartbeat Frequency

Maximum Number of Call

Lines

Fax Mode

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 Heartbeat Detection" 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.

Select Fax mode. The default setting is “None”.

None: Disable Fax.

Fax Detect: Fax signal from the user/trunk during the call can be detected and the received Fax will be sent to the Email address configured for this extension. If no Email address can be found for the user, the Fax will be sent to the default Email address configured in

Fax setting page under Web GUICall FeaturesFax/T.38.

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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 associated 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Extension/TrunkVoIP 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 CEO's direct line.

5.

If extension number need to be appended to the DID number click on “Add Extension”.

6.

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.

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Figure 104: DOD extension selection

7.

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 105: Edit DOD

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SLA STATION

The UCM6100 supports SLA (Shared Line Appearance) that allows mapping the key with LED on a multiline 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 trunks on the UCM6100. Users could configure the phone’s line key with BLF mode on the MPK to monitor the analog trunk status or press the line key to pick up call.

Create/Edit SLA Station

SLA Station can be configured on Web GUIExtension/TrunkSLA Station.

• Click on

• Click on

Figure 106: SLA Station

to add a SLA Station.

to edit the SLA Station. Following table shows SLA Station configuration parameters.

• Click on

to delete the SLA Station.

Station Name

Station

Available SLA Trunks

Selected SLA Trunks

Table 58: 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

Ring Timeout

Configure the time (in seconds) to ring the station before the call is considered unanswered. No timeout is set by default. If set to 0, there will be no timeout.

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Hold Access

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 GUIExtension/TrunkAnalog Trunks page. Create an analog trunk or edit the existing one. 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

GUIExtension/TrunkSLA Station page.

Figure 107: Enable SLA Mode for Analog Trunk

Click on “Save”. The analog trunk will be listed with trunk mode “SLA”.

Figure 108: Analog Trunk with SLA Mode Enabled

2.

On the UCM6100, go to Web GUIExtension/TrunkSLA Station page, click on “Create New SLA

Station”. Please refer to section

[Create/Edit SLA Station]

for the configuration parameters. Users can create one or more SLA stations to monitor the analog trunk. The following figure shows two stations,

1002 and 1005, are configured to be associated with SLA trunk “fxo1”.

Figure 109: 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 110: SLA Example - MPK Configuration

Now the SLA station is ready to use. The following functions can be achieved by this configuration.

Making an outbound call from the station/extension, using LINE key

When the extension is in idle state, pressing the line key for this extension on the phone to off hook.

Then dial the station’s extension number, for example, dial 1002 on phone 1 (or dial 1005 on phone 2), to hear the dial tone. Then the users could dial external number for the outbound call.

Making an outbound call from the station/extension, using BLF key

When the extension is in idle state, pressing the MPK and users could dial external numbers directly.

Answering call using LINE key

When the station is ringing, pressing the LINE key to answer the incoming call.

Barging-in active call using BLF key

When there is an active call between an SLA station and an external number using the SLA trunk, other

SLA stations monitoring the same trunk could join the call by pressing the BLF key if “Barge Allowed” is enabled for the analog trunk.

Hold/UnHold using BLF key

If the external line is previously put on hold by an SLA station, another station that monitors the same

SLA trunk could UnHold the call by pressing the BLF key if “Hold Access” is set to “open” on the analog trunk and the SLA station.

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CALL ROUTES

Outbound Routes

In the following sections, we will discuss the steps and parameters used to configure and manage outbound rules in UCM6100, these rules are the regulating points for all external outgoing calls initiated by the UCM through all types of trunks: SIP and Analog.

Configuring Outbound Routes

In the UCM6100, an outgoing calling rule pairs a dialed number pattern with a trunk used to dial out. 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Extension/TrunkOutbound Routes to add and edit outbound rules.

• Click on

to add a new outbound route.

• Click on

to edit the outbound route.

• Click on

to delete the outbound route.

On the UCM6100, the outbound route priority is based on “Best matching pattern”. For example, the

UCM6100 has outbound route A with pattern 1xxx and outbound route B with pattern 10xx configured.

When dialing 1000 for outbound call, outbound route B will always be used first. This is because pattern

10xx is a better match than pattern 1xxx. Only when there are multiple outbound routes with the same pattern configured.

Calling Rule Name

Pattern

Table 59: Outbound Route Configuration Parameters

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.

Notes:

▪ Multiple patterns can be used. Each pattern should be entered in new line.

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Disable This Route

Call Duration Limit

Maximum Call Duration

Warning Time

Warning Repeat Interval

PIN Groups

Password

Privilege Level

Enable Filter on Source

Caller ID

▪ Users can add comments to a dial plan by typing “/*” and “*/” before and after each comment respectively.

▪ Example:

_X.

_NNXXNXXXXX /* 10-digit long distance */

_818X. /* Any number with leading 818 */

After creating the outbound route, users can choose to enable and disable it. If the route is disabled, it will not take effect anymore. However, the route settings will remain in UCM. Users can enable it again when it’s needed.

Enable to set the maximum duration for the call using this outbound route.

Configure the maximum duration of the call (in sec). The default is 0s.

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.

If selected, the Password, Privilege Level and Enable Filter on Source

Caller ID will not take effect.

Configure the password for users to use this rule when making calls.

Select privilege level for the outbound rule.

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

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

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

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

Disable: The default setting is "Disable". If selected, only the matched source caller ID will be allowed to use this outbound route.

Please be aware of the potential security risks when using "Internal" level, which means all users can use this outbound rule to dial out from the trunk.

When enabled, users could specify extensions allowed to use this outbound route. "Privilege Level" is automatically disabled if using "Enable

Filter on Source Caller ID".

The following two methods can be used at the same time to define the extensions as the source caller ID.

1.

Select available extensions/extension groups from the left to the right. This allows users to specify arbitrary single extensions available in the PBX.

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2.

Custom Dynamic Route: define the pattern for the source caller ID.

This allows users to define extension range matching a pattern.

• 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.

Note: Multiple patterns can be used. Patterns should be separated by comma “,”. Example: _X. , _NNXXNXXXXX , _818X.

Send This Call Through Trunk

Use Trunk

Select the trunk for this outbound rule.

Strip

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.

Prepend

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

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

Use Failover Trunk

Failover Trunk

Strip

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.

UCM61XX support up to 10 failover trunks.

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.

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Prepend

Time Condition

Time Condition

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.

Users could customize holiday time, office time or a specified time to allow the outbound route to be used.

Outbound Blacklist

The UCM6100 allows users to configure blacklist for outbound routes. If the dialing number matches the blacklist numbers or patterns, the outbound call will not be allowed. The outbound blacklist can be configured under UCM Web GUIExtension/TrunkOutbound RoutesOutbound Blacklist.

Users can configure numbers, patterns or select country code to add in the blacklist. Please note that the blacklist settings apply to all outbound routes.

Figure 111: Outbound Blacklist

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PIN Groups

The UCM6100 supports pin group. Once this feature is configured, users can apply pin group to specific outbound routes. When placing a call on pin protected outbound routes, caller will be asked to input his pin number for authentication, this feature can be found under Extension/TrunkOutbound RoutesPIN

Groups.

Name

Record In CDR

PIN Number

PIN Name

Table 60: PIN Group

Specify the name of the group

Specify whether to enable/disable record in CDR

Specify the code that will asked once dialing via a trunk

Specify the name of the PIN

Once user click on the following figure pop’s up for configuring the new Pin group.

Figure 112: Create New PIN Group

Once PIN Groups and members Created it should look like the following:

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Figure 113: PIN members

If PIN group is enabled on outbound route level, password, privilege level and enable filter on source caller ID will be disabled.

Figure 114: Outbound PIN

If PIN group CDR is enabled, the call with pin group information will be displayed as part of CDR under

Account Code field.

Figure 115: CDR Record

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-

Importing PIN Groups from CSV files:

User can also import PIN Groups by uploading CSV files for each group. To do this:

1.

Navigate to Extension/TrunkOutbound RoutesPIN Groups and click on the “Choose file to upload” button.

Figure 116: Importing PIN Groups from CSV files

2.

Select the CSV file to upload. Incorrect file formats and improperly formatted CSV files will result in error messages such as the one below

:

Figure 117: Incorrect CSV File

3.

To ensure a successful import, please follow the format in the sample image below

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Figure 118: CSV File Format

• The top-left value (A1) is the PIN Group name. In this case, it is “ALPHA”.

• Row 2 contains the labels for the modifiable fields: pin and pin_name. These values should not be changed and will cause an upload error otherwise.

• Rows 3+ contain the user-defined values with Column A holding the PINs and Column B holding the PIN names. PIN values must consist of at least four digits.

• Once the file is successfully uploaded, the entry will be added to the list of PIN Groups.

Figure 119: CSV File Successful Upload

Inbound Routes

Inbound rules configuration is mandatory in order to set and manage the UCM’s behavior when receiving incoming calls from external world (Service provider, or other peer telephony systems).

In order to create and set the different parameters for inbound rules, users can follow the steps described on the following sections.

First, Inbound routes can be configured via Web GUIExtension/TrunkInbound Routes.

• Click on

to add a new inbound route.

• Click on "Blacklist" to configure blacklist for all inbound routes.

• Click on

to edit the inbound route.

• Click on

to delete the inbound route.

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Inbound Rule Configurations

Trunks

Pattern

Table 61: Inbound Rule Configuration Parameters

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, Pattern and CallerID

Pattern. The first part is used to specify the dialed number while 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 can call in or call out. For example, pattern '_2XXX/1234' means the only extension with the caller ID '1234' can use this rule.

Notes:

▪ Multiple patterns can be used. Each pattern should be entered in new line.

▪ Users can add comments to a dial plan by typing “/*” and “*/” before and after each comment respectively.

▪ Example:

Pattern CallerID Pattern

_X.

_NNXXNXXXXX /* 10-digit long distance */

_818X. /* Any number with leading 818 */

1000

1001

Disable This Route

After creating the inbound route, users can choose to enable and disable it. If the route is disabled, it will not take effect anymore. However, the route settings will remain in UCM. Users can enable it again when it’s needed.

Prepend Trunk Name

Prepend trunk name to display.

Prepend User Defined Name

Prepend a user defined name to display name.

Alert-Info

Inbound Multiple Mode

Configure the Alert-Info, when UCM6100 receives an INVITE request, the

Alert-Info header field specifies an alternative ring tone to the UAS.

Multiple mode allows user to switch between destinations of the inbound rule by feature codes. Configure related feature codes as described in

[Inbound Route: Multiple Mode]. If this option is enabled, user can use

feature code to switch between different destinations.

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Dial Trunk

Privilege Level

DID Destination

Default Destination

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.

Disable: Only the selected Extensions or Extension Groups are allowed to use this rule, when enabled Filter on Source Caller ID.

Internal: The lowest level required. All users are allowed to use this rule, check this level might be risky for security purpose.

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

National: Users with National or International Level can use this rule.

International: The highest level required. Only users with international level are allowed to use this rule.

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

• All

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

• External Number

• By DID

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Strip

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.

Configure the number of digits to be prepended to an inbound DID pattern, with strip taking precedence over prepend.

Prepend

Time Condition

Time Conditions

Destination

Select the time condition for the inbound rule.

Select the destination for the inbound call during specified time condition.

Inbound Route: Prepend Example

UCM6100 now allows user to prepend digits to an inbound DID pattern, with strip taking precedence over prepend. With the ability to prepend digits in inbound route DID pattern, user no longer needs to create multiple routes for the same trunk in order to route calls to different extensions.

Figure 120: Inbound Route feature: Prepend

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The following example demonstrates the process,

1.

If Trunk provides a DID pattern of 18005251163.

2.

If Strip is set to 8, UCM6100 will strip the first 8 digits.

3.

If Prepend is set to 2, UCM6100 will prepend 2 to the stripped number, now the number become 2163.

4.

UCM6100 will now forward the incoming call to extension 2163.

Inbound Route: Multiple Mode

In the UCM6100, the user can configure inbound route to enable multiple mode to switch between different destinations. The inbound multiple mode can be enabled under Inbound Route settings as shown on the following figure.

Figure 121: Inbound Route - Multiple Mode

When Multiple Mode is enabled for the inbound route, the user can configure a “Default Destination” and a

“Mode 1” destination for this route. By default, the call coming into this inbound route will be routed to the default destination.

SIP end devices that have registered on the UCM6100 can dial feature code *62 to switch to inbound route

“Mode 1” and dial feature code *61 to switch back to “Default Destination”. Switching between different mode can be easily done without Web GUI login.

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For example, the customer service hotline destination has to be set to a different IVR after 7PM. The user can dial *62 to switch to “Mode 1” with that IVR set as the destination before off work.

To customize feature codes for “Default Mode” and “Mode 1”, click on under “Inbound Routes” page, check “Enable Inbound Multiple Mode” option and change “Inbound Default

Mode” and “Inbound Mode 1” values (By default, *61 and *62 respectively).

Figure 122: Inbound Route - Multiple Mode Feature Codes

Fax Intelligent Route

The UCM6100 can automatically detect Fax and phone signal coming from the FXO port, and then forward

Fax or phone signal to the right destination. For example, when a regular phone call is coming, the

UCM6100 will be able to detect the phone signal and forward it through the correct inbound route to the destination; if Fax signal is coming, the UCM6100 will be able to forward it to the FXS extension where the

Fax machine is connected.

Fax with Two Media

The UCM6100 supports Fax re-INVITE with multiple codec negotiation. If a Fax re-INVITE contains both

T.38 and PCMA/PCMU codec, UCM6100 will choose T.38 codec over PCMA/PCMU. This feature is added since firmware 1.0.10.x. (In previous firmware version, i.e., 1.0.9.x or earlier, multiple codec in the re-INVITE is prohibited and will be dropped by UCM6100).

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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".

• 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 also be added as a Blacklist Number.

to add to the list. Anonymous can

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

.

Figure 123: Blacklist Configuration Parameters

• To add blacklist number in batch, click on “choose file to upload” to upload blacklist file in csv format.

Below figure shows an example of supported CSV file format.

Figure 124: Blacklist csv File

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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Call FeaturesFeature Codes.

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CONFERENCE ROOM

The UCM6100 supports conference bridging allowing multiple rooms to be used at the same time:

• UCM6102/6104 supports up to 3 conference rooms allowing up to 25 simultaneous PSTN or IP participants.

UCM6108/6116 supports up to 6 conference rooms allowing up to 32 simultaneous PSTN or IP participants.

The conference room configurations can be accessed under Web GUICall FeaturesConference. In this page, users could create, edit, view, invite, manage the participants and delete conference rooms. The conference room status and conference call recordings (if recording is enabled) will be displayed in this web page as well.

Conference Room Configurations

• Click on "Create New Conference Room" to add a new conference room.

• Click on

to edit the conference room.

• Click on

to delete the conference room.

Table 62: Conference Room Configuration Parameters

Extension

Configure the conference number for the users to dial into the conference.

Password

Admin Password

When configured, the users who would like to join the conference call must enter this password before accessing the conference room.

Note:

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

• The password has to be at least 4 characters.

Configure the password to join the conference room 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 call.

Note:

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

• The password has to be at least 4 characters.

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Enable Caller Menu

Record Conference

Quiet Mode

Wait For Admin

Enable User Invite

Announce Callers

Public Mode

Play Hold Music

Music On Hold

Skip Authentication When

Inviting User via Trunk from

Web GUI

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

If enabled, the calls in this conference room 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".

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 GUIPBX SettingsMusic On

Hold.

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

Conference Settings contains the following options

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Enable Talk detection

DSP Talking Threshold

Table 63: Conference Settings

If enabled, the AMI will send the corresponding event when a user starts or ends talking.

The time in milliseconds of sound above what the dsp has established as base line silence for a user before a user is considered to be talking. This value affects several operations and should not be changed unless the impact on call quality is fully understood, the default value is 128.

DSP Silence Threshold

The time in milliseconds of sound falling within the what the dsp has established as base line silence before a user is considered to be silent. This value affects several operations and should not be changed unless the impact on call quality is fully understood, the default value is 2500.

Users can check the talking Caller IDs in conference control page (UCM WebUICall

FeaturesConference). The image will move up and down when the user is talking.

Figure 125: Conference

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.

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Invite from Web GUI

For each conference bridge in UCM6100 Web GUICall FeaturesConference, 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 126: Conference Invitation From Web GUI

Invite by dialing 0 or 1 during conference call

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

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

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

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

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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.

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During The Conference

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

• Manage the conference call from Web GUI.

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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.

3

4

5

1

2

6

7

8

1

4

Table 64: Conference Caller IVR Menu

Conference Administrator IVR Menu

Mute/unmute yourself.

Lock/unlock the conference bridge.

Kick the last joined user from the conference.

Decrease the volume of the conference call.

Decrease your volume.

Increase the volume of the conference call.

Increase your volume.

More options:

• 1: List all users currently in the conference call.

• 2: Kick all non-Administrator participants from the conference call.

• 3: Mute/Unmute all non-Administrator participants from the conference call.

• 4: Record the conference call.

• 8: Exit the caller menu and return to the conference.

Conference User IVR Menu

Mute/unmute yourself.

Decrease the volume of the conference call.

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5

Decrease your volume.

6

Increase the volume of the conference call.

7

Increase your volume.

8

Exit the caller menu and return to the conference.

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Note:

When there is participant in the conference, the conference bridge configuration cannot be modified.

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Record Conference

The UCM6100 allows users to record the conference call and retrieve the recording from Web GUICall

FeaturesConferenceRecord Conference.

To record the conference call, when the conference bridge is in idle, enable "Record Conference" from the conference room 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 127: Conference Recording

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CONFERENCE SCHEDULE

Conference Schedule Configuration

Conference Schedule can be found under UCM6100 Web GUICall

FeaturesConferenceConference Schedule. Users can create, edit, view and delete a Conference

Schedule.

• Click on “Create New Conference Schedule” to add a new Conference Schedule.

• Click on the scheduled conference to edit or delete the event.

After the user configures UCM6100 with Google Service Settings [Google Service Settings Support] and

enables Google Calendar for Conference Schedule, the conference schedule on the UCM6100 can be synchronized with Google Calendar for authorized Google account.

Table 65: Conference Schedule Parameters

Schedule Options

Conference Subject

Conference Room

Kick Time(m)

Description

Repeat

Schedule Time

Enable Google Calendar

Configure the name of the scheduled conference. Letters, digits, _ and

- are allowed.

Select a conference room for this scheduled conference.

Set kick time before conference starts. When kick time is reached, a warning prompt will be played for all attendees in the conference room. After 5 minutes, this conference room will be cleared and locked for the scheduled conference to begin.

Note: Kick Time cannot be less than 6 minutes in order to clear the conference room.

The description of scheduled conference.

Repeat interval of scheduled conference. By default, set to single event.

Configure the beginning date and duration of scheduled conference.

Note: Please pay attention to avoid time conflict on schedules in the same conference room.

Select this option to sync scheduled conference with Google

Calendar.

Note: Google Service Setting OAuth2.0 must be configured on the

UCM6100. Please refer to section [Google Service Settings

Support].

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Conference Administrator

Local Extension

Select the administrator of scheduled conference from selected extensions.

Note: “Public Mode” must be disabled from Conference Room

Options.

Select available extensions from the list to attend scheduled conference.

Remote Extension

Select available extensions from the remote peer PBX.

Note: “LDAP Sync” must be enabled on the UCM6100 in order to view remote extensions here.

Send Mail

Special Extension

Remote Conference

Conference Room Options

Enable this option to send Invitation mails to each member participating on the conference.

Add extensions that are not in the list (both local and remote list). If the user wishes to add the special extension, please match the pattern on the outbound route.

Invite a remote conference.

Password

Admin Password

Configure conference room password. Please note that if “Public

Mode” is enabled, this option is automatically disabled.

Configure the password to join as conference administrator. Please note that if “Public Mode” is enabled, this option is automatically disabled.

Enable Caller Menu

If this option is enabled, conference participants will be able to access conference bridge menu by pressing the * key.

Record Conference

Quiet Mode

Wait For Admin

If this option is enabled, conference call will be recorded in .wav format. The recorded file can be found from Conference page.

If this option is enabled, the notification tone or voice prompt for joining or leaving the conference won’t be played.

Note: Option “Quiet Mode” and option “Announce Caller” cannot be enabled at the same time.

If this option is enabled, the participants in the conference won’t be able to hear each other until conference administrator joins the conference.

Note: If “Quiet Mode” is enabled, voice prompt for this option won’t be played.

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Enable User Invite

Announce Callers

Public Mode

Play Hold Music

Skip Authentication When

Inviting Users via Trunk from Web GUI

If this option is enabled, the user can:

• Press ‘0’ to invite others to join the conference with invited party’s permission

• Press ‘1’ to invite without invited party’s permission

• Press ‘2’ to create a multi-conference bridge to another conference room

• Press ‘3’ to drop all current multi-conference bridges

Note: Conference Administrator is always allowed to access this menu.

If this option is enabled, when a participant joins the conference room, participant’s name will be announced to all members in the conference room.

Note: Option “Quiet Mode” and option “Announce Caller” cannot be enabled at the same time.

If this option is enabled, no authentication is required for entering the conference room.

Note: Please be aware of the potential security risks when turning on this option.

If this option is enabled, UCM6100 will play Hold Music while there is only one participant in the conference room or the conference is not yet started.

If this option is enabled, the invitation from Web GUI via a trunk with password won’t require authentication.

Note: Please be aware of the potential security risks when turning on this option.

Cleaner Options

Cleaner Options

Enable Conference

Schedules Cleaner

Conference Schedules

Clean Time

Clean Interval

If this option is enabled, conference schedules will be automatically cleaned as configured.

Enter the clean time (in hours). The valid range is from 0 to 23.

Enter the clean interval (in days). The valid range is from 1 to 30.

Show/hide Conference Schedule Table

Enable this option will allow Web GUI to display scheduled conference in Conference Schedule Table.

Please see figure below.

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Figure 128: Conference Schedule

Once the conference room is scheduled, at the kick time, all users will be removed from conference room and no extension is allowed to join the conference room anymore. At the scheduled conference time,

UCM6100 will send INVITE to the extensions that have been selected for conference.

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Note:

• Please make sure that outbound route is properly configured for remote extensions to join the conference.

• Once Kick Time is reached, Conference Schedule is locked and cannot be modified.

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IVR

Configure IVR

IVR configurations can be accessed under the UCM6100 Web GUICall FeaturesIVR. 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.

Figure 129: Create New IVR

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Basic Settings

Name

Extension

Dial Trunk

Permission

Dial Other Extensions

IVR Black/White List

Internal Black/White List

External Black/White List

Replace Caller ID

Alert-Info

Prompt

Table 66: 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 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.

This controls the destination that can be reached by the external caller via the inbound route. The available destinations are:

• Extension

• Conference

• Call Queue

• Ring Group

• Paging/Intercom Groups

• Voicemail Groups

• Fax Extension

• Dial By Name

• All

If enabled only numbers inside of the White List or outside of the Black List can be called from IVR.

Contain numbers, either of Black List or White List.

This feature can be used only when Dial Trunk is enabled, it contains external numbers allowed or denied calling from the IVR, the allowed format is the following:

Number1, number2, number3, ….

If enabled, the UCM will replace the caller display name with the IVR name the caller know whether the call is incoming from a direct extension or IVR.

When present in an INVITE request, the alert-Info header field specifies and alternative ring tone to the UAS.

Select an audio file to play as the welcome prompt for the IVR. Click on

"Prompt" to add additional audio file under web GUIPBX

SettingsVoice PromptCustom Prompt.

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Digit Timeout

Response Timeout

Response Timeout Prompt

Invalid Prompt

Response Timeout Repeat

Loops

Invalid Repeat Loops

Language

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 the timeout, a timeout prompt will be played. The default setting is 10s.

Select the prompt message to be played when timeout occurs.

Select the prompt message to be played when an invalid DTMF 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 SettingsVoice PromptLanguage Settings. 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

SettingsVoice PromptLanguage Settings.

Key Pressing Events

Key Press Event:

Press 0

Press 1

Press 2

Press 3

Press 4

Press 5

Press 6

Press 7

Press 8

Press 9

Press *

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

• Callback

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Timeout

Invalid

When exceeding the number of defined answer timeout, IVR will enter the configured event when timeout. If not configured, then it will Hangup.

Configure the destination when the Invalid Repeat Loop is done.

Figure 130: Key Pressing Events

Black/White List in IVR

In some scenarios, the IPPBX administrator needs to restrict the extensions that can be reached from IVR.

For example, the company CEO and directors prefer only receiving calls transferred by the secretary, some special extensions are used on IP surveillance end points which shouldn’t be reached from external calls via IVR for privacy reason. UCM has now added blacklist and whitelist in IVR settings for users to manage this.

To use this feature, log in UCM Web GUI and navigate to Call FeaturesIVRCreate/Edit IVR: IVR

Black/White List.

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• If the user selects “Blacklist Enable” and adds extension in the list, the extensions in the list will not be allowed to be reached via IVR.

• If the user selects “Whitelist Enable” and adds extension in the list, only the extensions in the list can be allowed to be reached via IVR.

Figure 131: Black/White List

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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 SettingsVoice PromptCustom Prompt page directly.

Figure 132: 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.

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LANGUAGE SETTINGS FOR VOICE PROMPT

The UCM6100 supports multiple languages in web GUI as well as system voice prompt. Currently, there are 16 languages supported in system voice prompt: English (United States), Arabic, Chinese, Dutch,

English (United Kingdom), French, German, Greek, Hebrew, Italian, Polish, Portuguese, Russian,

Spanish, Swedish and Turkish.

English (United States) and Chinese voice prompts are built in with the UCM6100 already. The other languages provided by Grandstream can be downloaded and installed from the UCM6100 web GUI directly.

Additionally, users could customize their own voice prompts, package them and upload to the UCM6100.

Language settings for voice prompt can be accessed under Web GUIPBX SettingsVoice

PromptLanguage Settings.

Download and Install Voice Prompt Package

To download and install voice prompt package in different languages from UCM6100 web GUI, click on

"Check Prompt List" button.

Figure 133: Language Settings for Voice Prompt

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A new dialog window of voice prompt package list will be displayed. Users can see the version number

(latest version available V.S. current installed version), package size and options to upgrade or download the language.

Figure 134: Voice Prompt Package List

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

Figure 135: New Voice Prompt Language Added

A new language option will be displayed after successfully installed. Users then could select it to apply in the UCM6100 system voice prompt or delete it from the UCM6100.

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Customize Specific Prompt

On the UCM6100, if the user needs to replace some specific customized prompt, the user can upload a single specific customized prompt from Web GUIPBX SettingsVoice PromptLanguage Settings instead of the entire language pack.

Figure 136: Upload Single Voice Prompt for Entire Language Pack

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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Call FeaturesVoicemail.

Figure 137: Voicemail Settings

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Max Greeting

Dial ‘0’ For Operator

Operator Extension

Max Messages Per Folder

Max Message Time

Min Effective Message Time

Announce Message Caller-ID

Announce Message Duration

Play Envelope

Play from Last

Allow User Review

Table 67: Voicemail Settings

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.

Specify the operator extension, which will be dialed when users press 0 to exit voicemail application. The operator extension can also be used in IVR.

Configure the maximum number of messages per folder in users’ voicemail. The valid range 10 to 1000. The default setting is 50.

Select the maximum duration of the voicemail message. The message will not be recorded if the duration exceeds the max message time. The default setting is 15 minutes. The available options are:

• 1 minute

• 2 minutes

• 5 minutes

• 15 minutes

• 30 minutes

• Unlimited

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 3s. The available options:

• No minimum

• 1 second

• 2 seconds

• 3 seconds

• 4 seconds

• 5 seconds

Note: Silence and noise duration are not counted in message time.

If enabled, the caller ID of the user who has left the message will be announced at the beginning of the voicemail message. The default setting 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, UCM will play from the voice message left most recently; if disabled, UCM will play from the earliest left voice message

If enabled, users can review the message following the IVR before sending the message out. The default setting is "No".

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Access Voicemail

If the voicemail is enabled for UCM6100 extensions, the users can dial the voicemail access feature code

(by default *98 or *97) to access the extension’s voicemail. The users will be prompted to enter the voicemail password and then can enter digits from the phone keypad to navigate in the IVR menu for different options.

Main Menu

1 – New messages

2 – Change folders

3 – Advanced options

0 – Mailbox options

Table 68: Voicemail IVR Menu

Sub Menu 1

3 - Advanced options

Sub Menu 2

1 - Send a reply

2 - Call the person who sent this message

3 - Hear the message envelop

4 - Leave a message

* - Return to the main menu

5 - Repeat the current message

7 - Delete this message

8 - Forward the message to another user

9 - Save

* - Help

# - Exit

0 - New messages

1 - Old messages

2 - Work messages

3 - Family messages

4 - Friend messages

# - Cancel

1 - Send a reply

2 - Call the person who sent this message

3 - Hear the message envelop

4 - Leave a message

* - Return to the main menu

1 - Record your unavailable message

2 - Record your busy message

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1 - Accept this recording

2 - Listen to it

3 - Re-record your message

1 - Accept this recording

2 - Listen to it

3 - Re-record your message

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3 - Record your name

4 - Record temporary greeting

5 - Change your password

* - Return to the main menu

1 - Accept this recording

2 - Listen to it

3 - Re-record your message

1 - Accept this recording

2 - Listen to it

3 - Re-record your message

Extension Voicemail Count

The UCM6100 provides an easy way to check the number of voicemail messages for each extension directly from UCM web GUI  Extension/Trunk  Extensions overview page.

Voicemail count (“Message” column) is displayed in the format Urgent / Total / Read.

Figure 138: Voicemail Count

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.

Attach Recordings to E-Mail

Table 69: Voicemail Email Settings

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

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Keep Recordings

Template For Voicemail

Emails

If enabled, voicemail will be stored in the UCM6100 after the email is sent.

The default setting is “Yes”.

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

The template variables are:

• \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

Figure 139: Voicemail Email Settings

Click on "Email Template" 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Call

FeaturesVoicemail Group. Click on "Create New Voicemail Group" to configure the group.

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Figure 140: Voicemail Group

Extension

Name

Voicemail Password

Email Address

Voicemail Group Mailboxes

Table 70: 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Call FeaturesRing Group.

Figure 141: Ring Group

• Click on

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

Selected LDAP Numbers

Ring Strategy

Table 71: 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 available remote users from the left side to the ring group member list on the right side. Click on to arrange the order. Note:

LDAP Sync must be enabled first.

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

Ring simultaneously.

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

Ring 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.

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Music On Hold

Custom Prompt

Ring Timeout on Each

Member

Auto Record

Replace Caller ID

Enable Destination

Secret

Email Address

Select the “Music On Hold” Class of this Ring Group, “Music On Hold” can be managed from the “Music On Hold” panel on the left.

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 SettingsVoice

PromptCustom Prompt, where users could record new prompt or upload prompt files.

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

Note:

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

If enabled, calls on this ring group will be automatically recorded. The default setting is No. The recording files can be accessed from web

GUICDRRecording Files.

If enabled, the UCM will replace the caller display name with the Ring

Group name the caller know whether the call is incoming from a direct extension or a Ring Group.

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 ring group 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Call

FeaturesVoicemailVoicemail Email Settings, the voicemail can be sent to the ring group's Email address as attachment.

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Figure 142: Ring Group Configuration

Remote Extension in Ring Group

Remote extensions from the peer trunk of a remote UCM6100 can be included in the ring group with local extension. An example of Ring Group with peer extensions is presented in the following:

1.

Creating SIP Peer Trunk between both UCM6100_A and UCM6100_B. SIP Trunk can be found under web GUIExtension/TrunkVoIP Trunks. Also, please configure their Inbound/Outbound routes accordingly.

2.

Click edit button in the menu , and check if Sync LDAP Enable is selected, this option will allow

UCM6100_A update remote LDAP server automatically from peer UCM6100_B. In addition, Sync

LDAP Password must match for UCM6100_A and UCM6100_B in order to sync LDAP contact automatically. Port number can be anything between 0~65535, and use the outbound rule created in step 1 for the LDAP Outbound Rule option.

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Figure 143: Sync LDAP Server option

3.

In case if LDAP server doesn’t sync automatically, user can manually sync LDAP server. Under VoIP

Trunks page, click sync button shown in the following figure to manually sync LDAP contacts from peer

UCM6100.

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Figure 144: Manually Sync LDAP Server

4.

Under Ring Groups setting page, click “Add”. Ring Groups can be found under Web GUICall

FeaturesRing Groups.

5.

If LDAP server is synced correctly, Available LDAP Numbers box will display available remote extensions that can be included in the current ring group. Please also make sure the extensions in the peer UCM6100 can be included into that UCM6100’s LDAP contact.

Figure 145: Ring Group Remote Extension

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PAGING AND INTERCOM GROUP

Paging and Intercom Group can be used to make an announcement over the speaker on a group of phones.

Targeted phones will answer immediately using speaker. The UCM6100 paging and intercom can be used via feature code to a single extension or a paging/intercom group. This section describes the configuration of paging/intercom group under Web GUICall FeaturesPaging/Intercom.

Configure Paging/Intercom Group

• Click on "Create New Paging/Intercom Group" to add paging/intercom group.

Figure 146: Paging/Intercom Group

Name

Extension

Type

Custom Prompt

Page/Intercom Group

Members

Table 72: 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

PBXInternal OptionsCustom 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 147: 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Call FeaturesFeature

Codes directly.

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CALL QUEUE

The UCM6100 supports call queue by using static agents or dynamic agents. Call Queue system can accept more calls than the available agents. Incoming calls will be held until next representative is available in the system. This section describes the configuration of call queue under Web GUICall FeaturesCall

Queue.

Configure Call Queue

Call queue settings can be accessed via Web GUICall FeaturesCall Queue.

Figure 148: Call Queue

UCM6100 supports custom prompt feature in call queue. This custom prompt will active after the caller waits for a period of time in the Queue. Then caller could choose to leave a message/ transfer to default extension or keep waiting in the queue.

To configure this feature, please go to UCM Web GUICall FeaturesCall QueueCreate New

Queue/Edit QueueQueue Optionsset Enable Destination to Enter Destination with Voice Prompt. Users could configure the wait time with Voice Prompt Cycle.

• Click on

to add call queue.

• Click on

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

• Click on

to delete the call queue.

Table 73: Call Queue Configuration Parameters

Basic Settings

Extension

Name

Strategy

Configure the call queue extension number.

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.

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Music On Hold

Max Queue Length

Wrapup Time

Retry Time

Ring Time

Auto Record

Max Wait Time

Destination

Destination Prompt

Cycle

Custom Prompt

Linear

Ring agents in the specified order.

Least Recent

Ring the agent who has been called the least recently.

Fewest Calls

Ring the agent with the fewest completed calls.

Random

Ring a random agent.

Round Robin

Ring the agents in Round Robin scheduling with memory.

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

SettingsMusic On Hold.

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.

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 10 seconds.

Configure the number of seconds to wait before ringing the next agent.

Configure the number of seconds an agent will ring before the call goes to the next agent. The default setting is 30 seconds.

If enabled, the calls on the call queue will be automatically recorded. The recording files can be accessed in Queue Recordings under Web GUICall

FeaturesCall Queue.

Configure the timeout after which users will be disconnected from the call queue.

The default setting is "0" which means unlimited.

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

Once Max Wait Time has been configured, select to which destination send the calls that have timed out. The default is to “Hang up” the call.

Configure the voice prompt cycle (in seconds) of the call queue. Once all agents are busy and the voice prompt will be played and you can press the appropriate key to transfer to failover destination.

When playing a custom prompt, press 1 to transfer to failover destination.

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Destination

Select failover destination to send callers after pressing 1 upon hearing the custom prompt.

Advanced Settings

-

Virtual Queue

-

Position

Announcement

-

Queue Chairman

Refer to Call Center Settings & Enhancement section for detailed information

about these features.

Enable Agent Login

Leave When Empty

Enables agent login/logout feature for static agents (supported only on GXP21XX phones with fw higher than 1.0.9.18).

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".

Dial in Empty Queue

Yes

Callers can always dial into a call queue.

No

Callers cannot dial into a queue if all agents are paused or invalid.

Report Hold Time

Replace Display Name

Enable Feature Codes

Dynamic Login

Password

Alert-Info

Strict

Callers cannot dial into a queue if the agents are paused, invalid or unavailable.

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, the UCM will replace the caller display name with the Call Queue name so that the caller knows the call is incoming from a Call Queue.

Enable feature codes option for call queue. For example, *83 is used for “Agent

Pause”

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

Configure the call destination for the call to be routed to if no agent in this call queue answers the call.

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Agents

Agents

Go to “Agents” Tab and 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 choose. And use UP and Down arrow to select the order of the agent within the call queue.

Static Agents limitation:

To guarantee a high level of audio quality with the call queue feature, UCMs will limit the number of static agents allowed to be assigned depending on the UCM model used. If the user attempts to configure the number of static agents to be more than the maximum allowed number, a warning message will appear.

Figure 149: Static Agents limitation

The following table lists the maximum number of static agents for each UCM model:

Table 74: Static Agents limitation

UCM Model

UCM6102

UCM6202

UCM6104

UCM6204

UCM6108 and UCM6116

UCM6208

UCM6510

Max Static Agents in Call Queue

22

23

33

34

45

75

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

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

Figure 150: 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Call

FeaturesFeature Codes. The default feature code is *83 for "Agent Pause" and *84 for "Agent

Unpause".

• Queue recordings are shown on the Call Queue page under “Queue Recordings” Tab. 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.

Call Center Settings & Enhancement

UCM supports light weight call center features including virtual queue and position announcement, allowing the callers to know their position on the call queue and giving them the option to either stay on the line waiting for their turn or activate a callback which will be initiated by the UCM one an agent is free.

In order to configure call center features, press on an existing call queue and go under the advanced settings tab. Following parameters are available:

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Table 75: Call Center Parameters

Enable Virtual Queue

Enable virtual queue to activate call center features.

Virtual Queue Period

Configure the time in (s) after which the virtual queue will take effect and the menu will be presented to the caller to choose an option. Default is 20s.

Virtual Queue Mode

Offered to caller after timeout: After the virtual queue period passes, the caller will enter the virtual call queue and be presented with a menu to choose an option, the choices are summarized below:

• Press * to set current number as callback number.

• Press 0 to set a callback number different than current caller number.

• Press # to keep waiting on the call queue.

Triggered on user request: on this mode, the callers can activate the virtual queue by pressing 2, then they will be presented with the menu to choose an option as below:

• Press * to set current number as callback number.

• Press 0 to set a callback number different than current caller number.

• Press # to keep waiting on the call queue.

Virtual Queue

Outbound Prefix

Enable Position

Announcement

Position

Announcement

Interval

CTI Chairman

System will add this prefix to dialed numbers when calling back users.

Enable the announcement of the caller’s position periodically.

Configure the period of time in (s) during which the UCM will announce the caller’s position in the call queue.

Enable Agent Login

Select the extension to act as chairman of the queue (monitoring).

When enabled, statics agents can conveniently log in and out of a queue by configuring a programmable key on their phones as a shortcut.

Notes:

✓ This feature is currently available only for GXP21xx phones on firmware

1.0.9.18 or greater.

✓ After enabling the feature, users need to set the option on GXP21XX phone under “AccountSIP SettingsAdvanced FeaturesSpecial Feature” to “UCM Call Center”. A softkey labeled “UCM-CC” will appear on the bottom of the phone’s screen.

✓ When this option is enabled, dynamic agent login will be no longer supported.

Queue Auto fill enhancement:

In previous UCM firmware, the call queue had a serial type behavior in which the queue will make all the callers waiting in a queue, even if there is more than one available member, until the head caller is connected to the member to get to. The next waiting caller in line then becomes the head caller, and they

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are then connected with the next available member, and all available members and waiting callers keep waiting during this time.

Starting from 1.0.14.x, the waiting callers are connecting with available members in a parallel fashion until there are no more available members or no more waiting callers.

For example, in a call queue with linear method, if there are two available agents, when two callers call in the queue at the same time, UCM will assign the two callers to each of the two available agents at the same time, rather than assigning the second caller to second available agent after the first agent answers the call from the first caller.

Queue Statistics

Along with call center features, users can also gather detailed call queue statistics allowing them to make better changes/decision to manage better the call distribution and handling based on time, agent and queue.

To access call queue statistics, go to web GUICall FeaturesCall Queue and click on “Call Queue

Statistics”, the following page will be displayed:

Figure 151: Call Queue Statistics

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Select the time interval along with the queue(s) and agent(s) to get detailed statistics, then press on save button to get the information to be displayed (call distribution by agent, virtual queue distribution by hour...).

Switchboard

Switchboard is a web GUI tool for call queue monitoring and management, admin can access to it from the menu Call FeaturesCall Queue then press « Switchboard ».

Following page will be displayed:

Figure 152: Call Queue Switchboard

The table below gives a brief description for the main menus:

Waiting

Proceeding

Agents

Table 76: Switchboard Parameters

This menu shows the current waiting calls along with the caller id and the option to hang-up call by pressing on the button.

Shows the current established calls along with the caller id and the callee (agent) as well as the option to hang-up, transfer, add conference or barge-in the call using the call option buttons.

Displays the list of agents in the queue and the extension status (idle, ringing, in use or unavailable) along with some basic call statistics and agent’s mode (static or dynamic).

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There are three different privilege levels for Call Queue management from the switchboard: Super Admin,

Queue Chairman, and Queue Agent.

Super Admin - Default admin of the UCM. Call queue privileges include being able to view and edit all queue agents, monitor and execute actions for incoming and ongoing calls for each extension in

Switchboard, and generate Call Queue reports to track performance.

Queue Chairman - User appointed by Super Admin to monitor and manage an assigned queue extension via Switchboard. The Queue Chairman can log into the UCM user portal with his extension number and assigned user password. To access the Switchboard, click on “Value-added Features” in the side menu and click on “Call Queue”. In the image below, User 1012 is the Queue Chairman appointed to manage Queue Extension 6500 and can see all the agents of the queue in the

Switchboard.

Queue Agent - User appointed by Super Admin to be a member of a queue extension. A queue agent can log into the UCM user portal with his extension number and assigned user password. To access the Switchboard, click on “Value-added Features” in the side menu and click on “Call Queue”. However, a queue agent can view and manage only his own calls and statistics, but not other agents’ in the queue extension. In the image below, User 1007 is a queue agent and can see only his own information in the

Switchboard.

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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Call FeaturesPickup Groups.

• Click on

to create a new pickup group.

• Click on

to edit the pickup group.

• Click on

to delete the pickup group.

Select extensions from the list on the left side to the right side.

Figure 153: Edit Pickup Group

Configure Pickup Feature Code

When picking up the call for the pickup group member, the user only needs to dial the pickup feature code.

It’s not necessary to add the extension number after the pickup feature code. The pickup feature code is configurable under Web GUICall FeaturesFeature Codes.

The default feature code for call pickup extension is *8, otherwise if the person intending to pick up the call knows the ringing extension they can use ** followed by the extension number in order to perform the call pickup operation.

The following figure shows where you can customize these features codes.

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Figure 154: Edit Pickup Feature Code

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MUSIC ON HOLD

Music On Hold settings can be accessed via Web GUIPBX SettingsMusic 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 155: 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 start uploading. Users can upload:

➢ Single files with 8KHz Mono Music file, or

➢ Music on hold files in a compressed package with .tar, .tar.gz and .tgz as the suffix. The file name can only be letters, digits or special characters -_

➢ the size for the uploaded file should be less than 30M, the compressed file will be applied to the entire MoH.

• Users could also download all the music on hold files from UCM. In the Music On Hold page, click on

and the file will be downloaded to your local PC.

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• Click on

next to the sound file to delete it from the selected Music On Hold Class.

• Select the sound files and click on

to delete all selected music on hold files.

The UCM6100 allows Users to select the Music on Hold file from WebGUI to play it. The UCM6100 will initiate a call to the selected extension and play this Music on Hold file once the call is answered.

Steps to play the music on hold file:

1.

Click on the button for the Music on Hold file.

2.

In the prompted window, select the extension to playback and click .

Figure 156: Play Custom Prompt

3.

The selected extension will ring.

4.

Answer the call to listen to the music playback.

Users could also record their own Music on hold to override an existing custom prompt, this can be done by following those steps:

1.

Click on .

2.

A prompt of confirmation will pop up, as shown below.

Figure 157: Information Prompt

3.

Click .

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4.

In the prompted window, select the extension to playback and click .

Figure 158: Record Custom Prompt

5.

Answer the call and start to record your new music on hold.

6.

Hangup the call and refresh Music On Hold page then you can listen to the new recorded file.

----------------------------------------------------------------------------------------------------------------------------- ---------------

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 SERVER

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Call

FeaturesFax/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.

• Click on

to edit the Fax extension.

• Click on

to delete the Fax extension.

Figure 159: Fax Settings

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Enable Error Correction

Mode

Maximum Transfer Rate

Minimum Transfer Rate

Table 77: FAX/T.38 Settings

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 concurrent fax that can be sent by UCM6100. Two mode

“Only” and “More” are supported.

Max Concurrent Sending Fax

Only

Under this mode, the UCM6100 allows only single user to send fax at a time.

Fax Queue Length

Fax Header Information

More

Under this mode, the UCM6100 supports multiple concurrent fax sending by the users.

By default, this option is set to “only”.

Configure the maximum length of Fax Queue from 6 to 10. The default setting is 6.

Adds fax header into the fax file.

Configure the Email address to send the received Fax to if user's Email address cannot be found.

Default Email Address

Template Variables

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 not be received from Email.

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

The template variables are:

• ${CALLERIDNUM}: Caller ID Number

• ${CALLERIDNAME}: Caller ID Name

• ${RECEIVEEXTEN}: The extension to receive the Fax

• ${FAXPAGES}: Number of pages in the Fax

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

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Receiving Fax

Sample Configuration to Receive Fax from PSTN Line

The following instructions describe 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Extension/Trunk page.

4.

Create or edit the analog trunk for Fax as below.

Fax Mode: Make sure "Fax Mode" option is set to "None".

Figure 160: Configure Analog Trunk without Fax Detection

5.

Go to UCM6100 web GUIExtension/TrunkExtensions page.

6.

Create or edit the extension for FXS port.

Analog Station: Select FXS port to be assigned to the extension. By default, it's set to "None".

• Once selected, analog related settings for this extension will show up in "Analog Settings" section.

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Figure 161: Configure Extension for Fax Machine: FXS Extension

Figure 162: Configure Extension for Fax Machine: Analog Settings

7.

Go to web GUIExtension/TrunkInbound 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.

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Figure 163: Configure Inbound Rule for Fax

Now the Fax configuration is done. When there is an incoming Fax calling to the PSTN number for the FXO port, it will send the Fax to the Fax machine.

Sample Configuration for Fax-To-Email

The following instructions describe 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Call FeaturesFax/T.38 page. Create a new Fax extension.

Figure 164: Create Fax Extension

3.

Go to UCM6100 web GUIExtension/TrunkAnalog Trunks page. Create a new analog trunk.

Please make sure "Fax Detection" is set to "No".

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4.

Go to UCM6100 web GUIExtension/TrunkInbound Routes page. Create a new inbound route and set the default destination to the Fax extension.

Figure 165: Inbound Route to Fax Extension

5.

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

Fax Sending

The UCM6100 supports sending Fax via Web GUI access. This feature can be found on Web GUIValue-

added FeaturesFax Sending page. In order to send fax, pre-setup for analog trunk and outbound route is required. Please refer to

[ANALOG TRUNKS], [VOIP TRUNKS]

and

[Outbound Routes]

sections for configuring analog trunk and outbound route.

After making sure analog trunk or VoIP Trunk is setup properly and UCM6100 can reach out to PSTN numbers via the trunk, on Fax Sending page, enter the fax number and upload the file to be faxed. Then click on “Send” to start. The progress of sending fax will be displayed in Web GUI. Users can also view the sending history is in the same web page.

The following figures show sample of fax sending procedure and the progress when sending files.

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Figure 166: Fax Sending in Web GUI

After that you can see the ongoing sending operation on the progress bar.

Figure 167: Fax Send Progress

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BUSY CAMP-ON

The UCM6100 supports busy camp-on/call completion feature that allows the PBX to camp on a called party and inform the caller as soon as the called party becomes available given the previous attempted call has failed.

The configuration and instructions on how to use busy camp-on/call completion feature can be found in the following guide: http://www.grandstream.com/sites/default/files/Resources/ucm6xxx_busy_camp_on_guide.pdf

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PRESENCE

UCM does support SIP presence feature which allows users to advertise their current availability status and willingness to receive calls, this way other users can use their phones in order to monitor the presence status of each user and decide whether to call them or not based on their advertised availability.

This feature is different than BLF which is mainly used to monitor the dialog status for each extension

(Ringing, Idle or Busy). Instead the SIP presence module gives more options for users to choose which state they want to put themselves in.

In order to configure the presence status of an extension from the web GUI, users can access the menu of configuration using one of the two following methods:

• From admin account, go under the menu Extension/TrunkExtensions and choose the desired extension to edit then navigate to the “Features” tab.

OR

• From the User Portal, go under the menu Basic InformationExtensions and navigate to the

Features tab to have the following options.

Figure 168: SIP Presence Configuration

Select which status to set from the presence status selection drop list, six options are available and below is a brief description of these states:

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Table 78: SIP Presence Status

Available

Away

Chat

Do Not Disturb

Custom Presence Status

Unavailable

The contact is online and can participate in conversations/phone calls.

The contact is currently away (ex: for lunch break).

The contact has limited conversation flexibility and can only be reached via chat.

The Contact is on DND (Do Not Disturb) mode.

Please enter the presence status for this mode on the web GUI.

The contact is unreachable for the moment, please try to contact later.

Another option to set the presence status and which is more practical is using the feature code from the user’s phone, once the user dials the feature code (default is *48), a prompt will be played to select which status they want to put themselves in by pressing the corresponding key.

The feature code can be enabled and customized from the Web GUICall FeaturesFeature Codes.

Figure 169: SIP Presence Feature Code

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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. Follow Me option can be found under extension settings page web GUIExtension/TrunkExtensions.

To configure follow me:

1.

Choose the extension and click on .

2.

Go to the Follow me tab to add destination numbers and enable the feature.

Figure 170: Edit Follow Me

3.

Click on to add local extensions or external numbers once set to be called after ringing the extension selected in the first step.

4.

Once created, it will be displayed on the follow me web page list. Click on to delete the Follow Me.

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The following table shows the Follow Me configuration parameters.

Enable

Skip Trunk Auth

Music On Hold Class

Confirm When Answering

Enable Destination

Default Destination

Follow Me Numbers

New Follow Me Number

Dialing Order

Table 79: Follow Me Settings

Configure to enable or disable Follow Me for this user.

If external number is added in the Follow Me, please make sure this option is enabled or the “Skip Trunk Auth” option of the extension is enabled, otherwise the external Follow Me number cannot be reached.

Configure the Music On Hold class that the caller would hear while tracking the user.

By default, it is enabled and user will be asked to press 1 to accept the call or to press 2 to reject the call after answering a Follow Me call.

If it is disabled, the Follow Me call will be established once after the user answers it.

When enabled, the call will be routed to the default destination if no one in the Follow Me extensions answers the call.

Configure the destination if no one in the Follow Me extensions answers the call. The available options are:

• Extension

• Voicemail

• Queues

• Ring Group

• Voicemail Group

• IVR

• External 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.

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” under web GUIExtension/TrunkExtensions page to enable or disable the options listed in the following table.

Table 80: 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, the 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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SPEED DIAL

The UCM6100 supports Speed Dial feature that allows users to call a certain destination by pressing one or four digits on the keypad allowing 6561 combinations to be configured. This creates a system-wide speed dial access for all the extensions on the UCM6100.

To enable Speed Dial, on the UCM6100 Web GUI, go to page Web GUICall FeaturesSpeed Dial.

User should first click on . Then decide from one digit up to four digits combination used for

Speed Dial and 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 171: Speed Dial Destinations

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DISA

In many situations, the user will find the need to access his own IP PBX resources but he is not physically near one of his extensions. However, he does have access to his own cell phone. In this case, we can use what is commonly known as DISA (Direct Inward System Access). Under this scenario, the user will be able to call from the outside, whether it’s using his cell phone, pay phone, regular PSTN, etc. After calling into

UCM6100, the user can then dial out via the SIP trunk or PSTN trunk connected to UCM6100 as it is an internal extension.

The UCM6100 supports DISA to be used in IVR or inbound route. Before using it, create new DISA under web GUICall FeaturesDISA.

• Click on "Create New IVR" to add a new DISA.

• Click on

to edit the DISA configuration.

• Click on

to delete the DISA.

Figure 172: Create New DISA

The following tables gives details description of all available options in order to configure and set the DISA feature on the UCM:

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Name

Password

Permission

Response Timeout

Digit Timeout

Allow Hangup

Replace Display Name

Table 81: 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 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.

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".

If enabled, the UCM will replace the caller display name with the DISA name.

Once successfully created, users can configure the inbound route destination as "DISA" or IVR key event as "DISA". When dialing into DISA, users will be prompted with password first. After entering the correct password, a second dial tone will be heard for the users to dial out.

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CALLBACK

Callback is mainly designed for users who often use their mobile phones to make long distance or international calls which may have high service charges. The callback feature provides an economic solution for reduce the cost from this.

The callback feature works as follows:

1.

Configure a new callback on the UCM6100.

2.

On the UCM6100, configure destination of the inbound route for analog trunk to callback.

3.

Save and apply the settings.

4.

The user calls the PSTN number of the UCM6100 using the mobile phone, which goes to callback destination as specified in the inbound route.

5.

Once the user hears the ringback tone from the mobile phone, hang up the call on the mobile phone.

6.

The UCM6100 will call back the user.

7.

The user answers the call.

8.

The call will be sent to DISA or IVR which directs the user to dial the destination number.

9.

The user will be connected to the destination number.

In this way, the calls are placed and connected through trunks on the UCM6100 instead of to the mobile phone directly. Therefore, the user will not be charged on mobile phone services for long distance or international calls.

To configure callback on the UCM6100, go to web GUICall FeaturesCallback page. Configuration parameters are listed in the following table.

Table 82: Callback Configuration Parameters

Name

CallerID Pattern

Outbound Prepend

Delay Before Callback

Destination

Configure a name to identify the Callback.

Configure the pattern of the callers allowed to use this callback. The caller who places the inbound call needs to have the CallerID match this pattern so that the caller can get callback after hanging up the call.

Note: If leaving as blank, all numbers are allowed to use this callback.

Configure the prepend digits to be added at before dialing the outside number. The number with prepended digits will be used to match the outbound route. ‘-’ is the connection character which will be ignored.

Configure the number of seconds delay before calling back the user.

Configure the destination which the callback will direct the caller to. Two destinations are available:

• IVR

• DISA

The caller can then enter the desired number to dial out via UCM trunks.

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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.

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Note:

On the Grandstream GXP series phones, the MPK supports "Call Park" mode, which can be used to park the call by configuring the MPK number as call park feature code (e.g., 700). MPK "Call Park" mode can also be used to monitor and pickup parked call if the MPK number is configured as parking lot (e.g., 701).

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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 are under web GUICall FeaturesEvent List.

• Click on "Create New Event List" to add a new event list.

• Sort selected extensions manually in the Eventlist.

• Click on

to edit the event list configuration.

• Click on

to delete the event list.

URI

Local Extensions

Remote Extensions

Special Extensions

Table 83: 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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Figure 173: Create New Event List

Remote extension monitoring works on the UCM6100 via event list BLF, among Peer SIP trunks or Register

SIP trunks (register to each other). Therefore, please properly configure SIP trunks on the UCM6100 first before using remote BLF feature. Please note the SIP end points need support event list BLF in order to monitor remote extensions.

When an event list is created on the UCM6100 and remote extensions are added to the list, the UCM6100 will send out SIP SUBSCRIBE to the remote UCM6100 to obtain the remote extension status. When the

SIP end points register and subscribe 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.

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Note:

• To configure LDAP sync, please go to UCM6100 web GUIExtension/TrunkVoIP 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Call FeaturesDial By Name.

Figure 174: Create Dial by Name Group

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Figure 175: Configure Extension First Name and Last Name

1.

Name

Enter a Name to identify the Dial By Name group.

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Extension/TrunkExtensions 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.

4.

Prompt Wait Time

Configure “Prompt Wait Time” for Dial By Name feature. During Dial By Name call, the caller will need to input the first letters of First/Last name before this wait time is reached. Otherwise, timeout will occur and the call might hang up. The timeout range is between 3 and 60 seconds.

5.

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.

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6.

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.

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. If Dial By Name is set as a key pressing event for IVR, user could use ‘*’ to exit from Dial By

Name, then re-enter IVR and start a new event. The following example shows how to use this option.

Figure 176: Dial By Name Group In IVR Key Pressing Events

Figure 177: Dial By Name Group In Inbound Rule

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User Name Prompt Customization

Starting from fw 1.0.15.x, the Dial By Name feature can use the recorded name prompt of a user to announce his/her name assigned to the dialed extension. If no name prompt greeting exists, the name will be spelt out like in previous versions.

There are two ways to customize/set new user name prompt for an extension:

Upload User Name Prompt File from Web GUI

1.

First, Users should have a pre-recorded file respecting the following format:

• PCM encoded / 16 bits / 8000Hz mono.

• In “.GSM” or “.WAV” format.

• File size under 5M.

• Filename must be set as the extension number. For example, the recorded file name 1000.wav will be used for extension 1000.

2.

Go under web GUI PBX SettingsVoice Prompt User Name Prompt and click on

button.

3.

Select the recorded file to upload it and press Save and Apply Settings.

• Click on

to record again the user name prompt.

• Click on to play recorded user name prompt.

• Select user name prompts and press using the button

to delete specific file or select multiple files for deletion

.

Record User Name via Voicemail Menu

The second option to record user name is using voicemail menu, please follow below steps:

• Dial *98 to access the voicemail

• After entering the desired extension and voicemail password, dial “0” to enter the recordings menu and then “3” to record a name.

Another option is that each user can record their own name by following below steps:

• The user dials *97 to access his/her voicemail

• After entering the voicemail password, the user can press “0” to enter the recordings menu and then “3” to record his name.

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ACTIVE CALLS AND MONITOR

The active calls on the UCM6100 are displayed in Web GUISystem StatusActive Calls page. Users can monitor the status, hang up the call as well as barge in the active calls in real time manner.

Active Calls Status

To view the status of active calls, navigate to web GUISystem StatusActive Calls. The following figure shows extension 1002 is calling 1004. 1004 is ringing.

Figure 178: StatusPBX StatusActive Calls - Ringing

The following figure shows the call between 1002 and 1003 is established.

Figure 179: StatusPBX StatusActive Calls – Call Established

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The green color of the active call means the connection of call time is less than half an hour. It means this call is normal.

Figure 180: call connection less than half hour

The yellow color of the active call means the connection of call time is greater than half an hour but less than one hour. It means this call is a bit long.

Figure 181: call connection between half an hour and one hour

The red color of the active call means the connection of call time is more than one hour. It means this call could be abnormal.

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Figure 182: Call Connection More Than One Hour

Hang Up Active Calls

To hang up an active call, click on

to hang up all active calls.

Call Monitor

icon in the active call dialog. Users can also click on

During an active call, click on icon and the monitor dialog will pop up.

Figure 183: Configure to Monitor an Active Call

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In the “Monitor” dialog, configure the following to monitor an active call:

1.

Enter an available extension for “Monitor’s Extension” which will be used to monitor the active call.

2.

“Monitored Extension” must be one of the parties in the active call to be monitored.

3.

Select spy mode. There are three options in “Spy Mode”.

Listen

In “Listen” mode, the extension monitoring the call can hear both parties in the active call but the audio of the user on this extension will not be heard by either party in the monitored active call.

Whisper

In “Whisper” mode, the extension monitoring the call can hear both parties in the active call. The user on this extension can only talk to the selected monitored extension and he/she will not be heard by the other party in the active call. This can be usually used to supervise calls.

Barge

In “Barge” mode, the extension monitoring the call can talk to both parties in the active call. The call will be established similar to three-way conference.

4.

Enable or disable “Require Confirmation” option. If enabled, the confirmation of the invited monitor’s extension is required before the active call can be monitored. This option can be used to avoid adding participant who has auto-answer configured or call forwarded to voicemail.

5.

Click on “Add”. An INVITE will be sent to the monitor’s extension. The monitor can answer the call and start monitoring. If “Require Confirmation” is enabled, the user will be asked to confirm to monitor the call.

Another way to monitor active calls is to dial the corresponding feature codes from an extension. Please refer to

[Table 84: UCM6100 Feature Codes]

and

[Enable Spy]

section for instructions.

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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 84: UCM6100 Feature Codes

Feature Maps

Blind Transfer

• 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.

Attended Transfer

Seamless Transfer

• 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: *44 (Disabled by default).

• Seamless Transfer allows user to perform blind transfer using

UCM feature code without having music on hold presented during the transfer process, it minimizes the interruption during transfer, making the process smooth and simple.

• During an active call use the feature code (*44 by default) followed by the number you want to transfer to in order to perform the seamless transfer.

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Disconnect

Call Park

Audio Mix Record

• 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.

• 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

Call Forward Busy Deactivate

• Default Code: *90

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

• Default Code: *91

Call Forward No Answer Activate

• Default Code: *92

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

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Call Forward No Answer

Deactivate

Call Forward Unconditional

Activate

Call Forward Unconditional

Deactivate

Feature Misc

Feature Code Digits Timeout

Call Park

Parked Lots

Use Parklot as Extension

Parking Timeout (s)

• Default Code: *93

• Default Code: *72

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

• Default Code: *73

• Default Setting: 1000

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

• Default Extension: 700

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

• Default Extension: 701-720

• These are the extensions where the calls will be parked, i.e., parking lots that the parked calls can be retrieved.

• If checked, the parking lot number can be used as extension.

The user can transfer the call to the parking lot number to park the call. Please note this parking lot number range might conflict with extension range.

• 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.

Select the Music on Hold Class.

Music On Hold Classes

Feature Codes

Voicemail Access Code

My Voicemail

Agent Pause

• 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.

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Agent Unpause

Paging Prefix

Intercom Prefix

Blacklist Add

Blacklist Remove

Call Pickup on Ringing

Pickup In-call

Pickup Extension

Direct Dial Voicemail Prefix

• 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: *45 (Disabled by default).

• If “Pickup In-call” feature is enabled, only the extensions added in “Allowed to seamless transfer” in the extension’s Seamless

Transfer Privilege Control List” can pick up the call.

• Default Code: *8

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

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

• Default Code: *

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

• For example, directly dial *5000 will have to call go into the extension 5000's voicemail. If the user would like to transfer the call to the extension 5000's voicemail, enter *5000 as the transfer target number.

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Direct Dial Mobile Phone Prefix

Call Completion Request

Call Completion Cancel

Enable Spy

Listen Spy

Whisper Spy

Barge Spy

Wakeup Service

PMS Wakeup Service

Update PMS Room Status

Presence Status

• Default Code: *88

• If you have the permission to call mobile phone number, use this prefix plus the extension number can dial the mobile phone number of this extension directly.

• Default Code: *11

• This code is for the user who wants to use Call Completion to complete a call.

• Default Code: *12

• This code is for the user who wants to cancel Call Completion request.

Check this box to enable spy feature codes. Disabled by default.

• Default Code: *54 (“Enable Spy” needs to be checked)

• This is the feature code to listen in on a call to monitor performance. Monitor’s line will be muted, and neither party will hear from the monitor’s extension.

• Default Code: *55 (“Enable Spy” needs to be checked)

• This is the feature code to speak to one side of the call (for example, whisper to employees to help them handle a call).

Only one side will be able to hear from the monitor’s extension.

• Default Code: *56 (“Enable Spy” needs to be checked)

• This is the feature code to join in on the call to assist both parties.

• Default Code: *36

• Dial this code to access UCM wakeup service, you can add, update, activate or deactivate wakeup service.

• Default Code: *35

• Dial this code to access UCM PMS wakeup service, you can add, update, activate or deactivate PMS wakeup service.

• Default Code: *23

• Use this code with maid code to update PMS room status.

Choose the status to set after hearing the prompt, for example: for maid 001 dial *23001 and then 1 after hearing the prompt.

• Dial this code to set the presence status of the extension.

• Possible options are:

1:"unavailable"

2:"available"

3:"away"

4:"chat

5:"dnd

6:"userdef"

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The UCM6100 also allows user to enable / disable specific feature code by one click. As shown below:

Figure 184: Enable/Disable Feature codes

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CDR. Click on to show and play the recording or click on to download the recording file.

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Figure 185: Download Recording File from CDR Page

The above recorded call's recording files are also listed under the UCM6100 web GUICDRRecording

Files.

Figure 186: Download Recording File from Recording Files Page

Call Park

The UCM6100 provides call park and call pickup features via feature code.

Park a Call

There are two feature codes that can be used to park the call.

• Feature MapsCall Park (Default code #72)

During an active call, press #72 and the call will be parked. Parking lot number (default range 701 to

720) will be announced after parking the call.

• 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 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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Enable Spy

If “Enable Spy” option is enabled, feature codes for Listen Spy, Whisper Spy and Barge Spy are available for users to dial from any extension to perform the corresponding actions.

Assume a call is on-going between extension A and extension B, user could dial the feature code from extension C to listen on their call (*54 by default), whisper to one side (*55 by default), or barge into the call

(*56 by default). Then the user will be asked to enter the number to call, which should be either side of the active call, extension A or B in this example.

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Caution:

“Enable Spy” allows any user to listen to any call by feature codes. This may result in the leakage of user privacy.

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PBX SETTINGS

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

SettingsGeneral Settings.

General Settings

Table 85: Internal Options/General

General Preferences

Global Outbound CID

Global Outbound CID Name

Ring Timeout

Call Duration Limit

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.

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.

Configure the maximum duration of call-blocking.

If enabled, users will hear voice prompt before recording is started or stopped. For example, before recording, the UCM6100 will play voice prompt "The call will be recorded". The default setting is "No".

Extension Preferences

Enforce Strong Passwords

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

Strong Password Rules:

1.

Password for voicemail, voicemail group, outbound route, DISA, call queue and conference requires non-repetitive and non-sequential digits, with a minimum length of 4 digits. Repetitive digits pattern (such as 0000, 1111, 1234, 2345, and etc), or common digits pattern (such as 111222, 321321 and etc) are not allowed to be configured as password.

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Enable Random Password

Enable Auto E-mail

Notification

Disable Extension Range

Extension Ranges

2.

Password for extension registration, web GUI admin login, LDAP and

LDAP sync requires alphanumeric characters containing at least two categories of the following, with a minimum length of 4 characters.

• Numeric digits

• Lowercase alphabet characters

• Uppercase alphabet characters

• Special characters

If enabled, random password will be generated when the extension is created. The default setting is "Yes". It is recommended to enable it for security purpose.

If enabled, UCM6100 will send Email notification to user automatically after editing extension settings or adding a new extension.

If set to "Yes", users could disable the extension range preconfigured/configured on the 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

GUIExtension/TrunkExtensions 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 extensions can be assigned on the UCM6100 to provision the end device.

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Voice Prompts Customization

• Conference Extensions: 6300-6399

• Ring Group Extensions: 6400-6499

• Queue Extensions: 6500-6599

• Voicemail Group Extensions: 6600-6699

• IVR Extensions: 7000-7100

• Dial By Name Extensions: 7101-7199

• Fax Extensions: 7200-8200

Record New Custom Prompt

Settings

In the UCM6100 Web GUIPBX SettingsVoice PromptCustom Prompt page, click on “Record New

Custom Prompt” and follow the steps below to record new IVR prompt.

Figure 187: Record New Custom Prompt

1.

Specify the IVR file name.

2.

Select the format (GSM or WAV) for the IVR prompt file to be recorded.

3.

Select the extension to receive the call from the UCM6100 to record the IVR prompt.

4.

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.

5.

Pick up the call from the extension and start the recording following the voice prompt.

6.

The recorded file will be listed in the IVR Prompt web page. Users could select to re-record, play or delete the recording.

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Upload Custom Prompt

If the user has a pre-recorded IVR prompt file, click on “Upload Custom Prompt” in Web GUIPBX

SettingsVoice PromptCustom 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 188: Upload Custom Prompt

Click on “choose file to upload” to start uploading. Once uploaded, the file will appear in the Custom Prompt web page.

Download All Custom Prompt

On the UCM61XX, the users can download all custom prompts from UCM Web GUI to local PC. To download all custom prompt, log in UCM Web GUI and navigate to PBX SettingsVoice

. The following window will pop up in order PromptCustom Prompt and click on to set a name for the downloaded file.

Figure 189: Download All Custom Prompt

Note: The downloaded file will have a .tar extension.

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PBX Settings/Jitter Buffer

Table 86: Internal Options/Jitter Buffer

SIP Jitter Buffer

Enable Jitter Buffer

Jitter Buffer Size

Max Jitter Buffer

Implementation

PBX Settings/RTP Settings

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".

RTP Start

RTP End

Strict RTP

RTP Checksums

ICE Support

Table 87: Internal Options/RTP Settings

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

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

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

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

Configure whether to support ICE. The default setting is enabled.

ICE is the integrated use of STUN and TURN structure to provide reliable

VoIP or video calls and media transmission, via a SIP request/ response model or multiple candidate endpoints exchanging IP addresses and ports, such as private addresses and TURN server address.

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STUN Server

Configure STUN server address. STUN protocol is a Client/Server and also a Request/Response protocol. It’s used to check the connectivity between the two terminals, such as maintaining a NAT binding entries keep-alive agreement. The default STUN Server is stun.ipvideotalk.com.

Valid format:

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

The default port number is 3478 if not specified.

PBX Settings/Payload

The UCM6100 payload type for audio codecs and video codes can be configured here.

Table 88: Internal Options/Payload

AAL2-G.726

Configure payload type for ADPCM (G.726, 32kbps, AAL2 codeword packing). The default setting is 112.

DTMF

G.721 Compatible

G.726

Configured payload type for DTMF. The default setting is 101.

Configure to enable/disable G.721 compatible. The default setting is Yes.

Configure the payload type for G.726 if "G.721 Compatible" is disabled.

The default setting is 111.

iLBC

H.264

H.263P

VP8

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.

Configure the payload type for VP8. The default settings is 108.

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IAX SETTINGS

The UCM6100 IAX global settings can be accessed via Web GUIPBX SettingsIAX Settings.

IAX Settings/General

Bind Port

Bind Address

IAX1 Compatibility

No Checksums

Delay Reject

ADSI

Music On Hold Interpret

Music On Hold Suggest

Bandwidth

Table 89: 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 feature. 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".

IAX Settings/Registration

Table 90: IAX Settings/Registration

IAX Registration Options

Min Reg Expire

Max Reg Expire

IAX Thread Count

IAX Max Thread Count

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

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

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

Configure the maximum number of IAX threads allowed. The default setting is 100.

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Auto Kill

Authentication Debugging

Codec Priority

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/Security

Call Token Optional

Max Call Numbers

Max Unvalidated Call

Numbers

Call Number Limits

IP or IP Range

Table 91: IAX Settings/Static Defense

Enter a single IP address (e.g., 11.11.11.11) or a range of IP addresses

(11.11.11.11/22.22.22.22) for which call token validation is not required.

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

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

Configure to limit the number of calls for a give IP address of IP range.

Enter the IP address (11.11.11.11) or a range of IP addresses

(11.11.11.11/22.22.22.22) to be considered for call number limits.

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SIP SETTINGS

The UCM6100 SIP global settings can be accessed via Web GUIPBX SettingsSIP Settings.

SIP Settings/General

Realm For Digest

Authentication

Bind UDP Port

Bind IP Address

Allow Guest Calls

Allow Transfer

MWI From

Enable Diversion Header

Table 92: 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.

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".

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.

If disabled, the UCM will not forward the diversion header.

SIP Settings/Misc

Table 93: 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

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".

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Reject Non-Matching INVITE

Allow callback when blind transfer fails.

Blind transfer timeout

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".

SDP Attribute Passthrough

Enable Attribute Passthrough

If enable, and if the service doesn’t know the attribute of

FEC/FECC/BFCP, then the attribute will be passthrough.

Early Media

Enable Use Final SDP

Blind Transfer

If enabled, call negotiation will use final response SDP.

If enabled, the UCM will callback to the transferrer when blind transfer fails

(reason of failure includes: busy and no answer).

Note:

This feature takes effect only on internal calls.

Configure the timeout in (s) for the transferrer waiting for the destination to answer. Default is 60s.

SIP Settings/Session Timer

Session Timers

Table 94: SIP Settings/Session Timer

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 another UA.

Refuse

Do not run session timer.

Session Expire

Min SE

Session Refresher

Configure the maximum session refresh interval (in seconds). The default setting is 1800.

Configure the minimum session refresh interval (in seconds). The default setting is 90.

Select the session refresher to be UAC or UAS. The default setting is

UAC.

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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

TLS Cert

TLS CA Cert

TLS CA List

Table 95: 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 selfsigned certificate, including server's public key. This file will be renamed as "TLS.ca" automatically.

Note:

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

This is the Certificate file (*.pem format only) used for TLS connections. It contains private key for client and signed certificate for the server. This file 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.

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SIP Settings/NAT

External Host

Use IP address in SDP

External TCP Port

External TLS Port

Local Network Address

Table 96: SIP Settings/NAT

Configure a static IP address and port (optional) used in outbound SIP messages if the UCM6100 is behind NAT. If it is a host name, it will only be looked up once.

If enabled, the SDP connection will use the IP address resolved from the external host.

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

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

Enable Relaxed DTMF

DTMF Mode

Table 97: 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.

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".

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RTP Timeout

RTP Hold Timeout

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.

RTP Keep-alive

100rel

Trust Remote Party ID

Send Remote Party ID

Generate In-Band Ringing

This feature can be used to avoid abnormal call drop when the remote provider requires RTP traffic during proceeding.

For example, when the call goes into voicemail and there is no RTP traffic sent out from UCM, configuring this option can avoid voicemail drop.

When configured, RTP keep-alive packet will be sent to remote party at the configured interval. If set to 0, RTP keep-alive is disabled.

Configure the 100rel setting on UCM6100.

The default setting is “Yes”.

Configure whether the Remote-Party-ID should be trusted.

The default setting is "No".

Configure whether the Remote-Party-ID should be sent or not.

The default setting is "No".

Configure whether the UCM6100 should generate inband ringing or not.

The default setting is "Never".

Yes: The UCM6100 will send 180 Ringing followed by 183 Session

Progress and in-band audio.

No: The UCM6100 will send 180 Ringing if 183 Session Progress has not been sent yet. If audio path is established already with 183 then send in-band ringing.

Never: Whenever ringing occurs, the UCM6100 will send 180 Ringing as long as 200OK has not been set yet. Inband ringing will not be generated even the end point device is not working properly.

Server User Agent

Configure the user agent string for the UCM6100.

Send Compact SIP Headers

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

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SIP Settings/SIP Trunk Prompt Tone

SIP Trunk Prompt Tone tab has been added to the UCM to help user choose which prompt will be played by the UCM during call failure, the following voice message responses have been added and can be set to play for 4XX, 5XX, and 6XX call failures:

• Default for 404 and 604 status codes: Your call can’t be completed as dialed. Please check the

number and dial again.

• Default for 5xx status codes: Server error. Please check your device.

• Default for 403 and 603 status codes: The call was rejected by the server. Please try again later.

• Default for all other status codes: All circuits are busy now. Please try again later.

Additionally, custom voice messages recorded and uploaded in PBX SettingsVoice PromptCustom

Prompt can be used for these failure responses instead of the default messages.

Figure 190: SIP Trunk Prompt Tone

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Transparent Call-Info header

UCM supports transparent call info header in order to integrate GDS door system with GXP21XX Color phones, the UCM will forward the call-info header to the phone in order to request the live view from GDS door system and give the option to open the door via softkey.

Figure 191: Transparent Call-Info

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INTERFACE SETTINGS

Analog Hardware

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 192: 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" and choose the detection algorithm, two algorithms exist (ERL,

Pr) for the UCM6100 to automatically detect the ACIM value. The detecting value will be automatically filled into the settings.

Figure 193: FXO Ports ACIM Settings

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PCMA Override

Boost Ringer

Fast Ringer

Low Power

Ring Detect

FXS MWI Mode

Tone Region

Advanced Settings

FXO Opermode

FXS Opermode

FXS TISS Override

Table 98: PBX Interface Settings

Select country to set the default tones for dial tone, busy tone, ring tone and etc to be sent from the FXS port. The default setting is "United States of America (USA)".

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

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

Operational Loop Current, and AC Impedance as predefined for your country's analog line characteristics. The default setting is "United 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 600Ω.

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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DAHDI Settings

When users encounter issues such as audio delay in outbound calls using the analog trunk, they can adjust

DAHDI settings on the UCM to attempt to lessen or resolve the issues.

Figure 194: DAHDI Settings

For the value of the option such as “32, half”:

The number in the option indicates the number of read/write buffers for TDM (DAHDI).

The “Half”, “Immediate” or “Full” option indicates the strategy when reading/writing data from buffer.

-

“Half”: Data will be read/written from buffer when half of the buffer is occupied with data.

-

“Immediate”: Read/write from buffer whenever there is data occupying the buffer.

-

“Full”: Data will be read/written from buffer when buffer is fully occupied with data.

Normally, DAHDI settings should be kept default and should be adjusted only when users encounter analog trunk/Fax-related issues.

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CTI SERVER

UCM does support CTI server capabilities which are designed to be a part of the CTI solution suite provided by Grandstream, including GXP21XX and GXP17XX enterprise IP phones along with GS Affinity app.

Mainly the UCM will by default listening on port TCP 8888 for the connections from GS affinity application in order to interact, modify and serve data requests by the application which includes setting call features for the connected extension as call forward and DND.

Users can change the listening port under the menu page, Web GUIValue-added FeaturesCTI Server as shown on below screenshot:

Figure 195: CTI Server Listening port

More information about GS affinity and CTI Support on Grandstream products series please refer to the following link: http://www.grandstream.com/sites/default/files/Resources/GS_Affinity_Guide.pdf

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ASTERISK MANAGER INTERFACE (RESTRICTED ACCESS)

The UCM6100 supports Asterisk Manager Interface (AMI) with restricted access. AMI allows a client program to connect to an Asterisk instance commands or read events over a TCP/IP stream. It’s particularly useful when the system admin tries to track the state of a telephony client inside Asterisk.

User could configure AMI parameters on UCM6100 web GUIValue-added FeaturesAMI. For details on how to use AMI on UCM6100, please refer to the following AMI guide: http://www.grandstream.com/sites/default/files/Resources/UCM_series_AMI_guide.pdf

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Warning:

Please do not enable AMI on the UCM6100 if it is placed on a public or untrusted network unless you have taken steps to protect the device from unauthorized access. It is crucial to understand that AMI access can allow AMI user to originate calls and the data exchanged via AMI is often very sensitive and private for your

UCM6100 system. Please be cautious when enabling AMI access on the UCM6100 and restrict the permission granted to the AMI user. By using AMI on UCM6100 you agree you understand and acknowledge the risks associated with this.

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PMS INTEGRATION

UCM6100 series support Hotel Property Management System PMS, including check-in/check-out services, wakeup calls, room status, Do Not Disturb which provide an ease of management for hotel applications.

This feature can be found on Web GUIValue-added FeaturesPMS.

Note: The PMS integration on UCM is currently supported only with one of the two following solutions.

HMobile PMS Connector

In this mode, the system can be divided into three parts:

• PMS (Property Management System)

• PMSI (Property Management System Interface)

• PBX

Grandstream UCM6XXX series have integrated HMobile Connect PMSI which supports a large variety of

PMS software providing following hospitality features: Check-in, Check-out, set Room Status, Wake-up call and more.

The following figure illustrates the communication flow between the PBX (Grandstream UCM6xxx Series) and PMS software, which is done through a middleware system (HMobile Connect) acting as interface between both parties.

PMS Software

PMSI Middleware

(HMobile Connect)

Grandstream UCM

Figure 196: UCM & PMS interaction

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Mitel PMS

In this mode, the system can be divided into two parts:

• PMS (Property Management System)

• PBX

Grandstream UCM6XXX series have integrated Mitel PMS providing following hospitality features: Checkin, Check-out, set Room Status, Wake-up call and more.

The following figure illustrates the communication flow between the PBX (Grandstream UCM6xxx Series) and PMS software (Mitel). The communication between both parties is direct with no middleware.

Mitel PMS Software

Grandstream UCM

Figure 197: UCM & PMS interaction

The PMS module built-in the UCM supports the following features based on each solution:

Feature

Check-In

Check-out

Wake-up Call

Name Change

Update

Set Credit

Set Station Restriction

Room Status

Room Move

Do Not Disturb

Mini Bar

MSG

Table 99: PMS Supported Features

Mitel

Hmobile

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Configuration

In order to use all PMS features please activate the feature code associated under “Call

FeaturesFeature Codes

-

Enable PMS

-

Update PMS Room Status

-

PMS Wake Up Service Activate

-

PMS Wake Up Service Deactivate

Basic Settings

On the UCM WebGUIValue-added FeaturesPMSBasic Settings” set the connection information for the

HMobile platform.

Field

PMS Module

Wake Up Prompt

PMS URL

UCM Port

Username

Password

PMS Features

Room Status

Table 100: PMS Basic Settings

Description

Users can select the desired PMS module from the drop-down list.

Hmobile.

Mitel.

Prompt used when answering the wakeup calls it can be customized from “PBX>Internal Options>Custom Prompt

Enter the PMS system URL

Enter the Port used by the PMS system

Enter the Username to connect to the PMS system

Enter the password to connect to the PMS system

User can create Rooms by clicking on , the following Figure will be displayed then.

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Figure 198: Create New Room

Click “Save” to create the new room, the fields above can be configured from the HMobile platform, once set the following screen will be shown:

Figure 199: Room Status

User can Create a batch of rooms as well by clicking on , the following window will pop up:

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Figure 200: Add batch rooms

Wakeup Service

In order to create a new Wakeup service, user can click on will pop up:

, the following window

Field

Room Number

Time

Action Status

Type

Figure 201: Create New Wake Up Service

Table 101: PMS Wake up Service

Description

Select the room number where to call

Set the time of the wakeup call

Show the status of the call:

Programmed: the call is scheduled for the time set

Cancelled: the call is canceled

Executed: the wakeup call is made

Single: The call will be made once on the specific time.

Daily: The call will be repeated every day on the specific time

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Once the call is made on the time specified, the following figure show the status of the wakeup call.

Figure 202: Wakeup Call executed

This call has been executed but has been rejected, that why we can see the “Busy” status.

Mini Bar

In order to create a new mini bar, click on

FeaturesPMSMini Bar, the following window will pop up:

under UCM webGUIValue-added

Figure 203: Create New Mini Bar

Code

Name

Prompt

Skip Maid and Password

Authentication

Enable Continuous Multi Goods

Billing

Table 102: Create New Mini Bar

Enter a non-existing extension number to be dialed when using the mini bar feature.

Enter a name for the mini bar.

Select the Prompt to play once connected to the mini bar.

If enabled, the default maid code will be 0000, no authentication is required. (Enter 0000 followed by # to access the consumer goods)

If enabled, please separate the goods’ codes by*.

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In order to create a new maid, click on

FeaturesPMSMini Bar, the following window will popup.

under UCM Web GUIValue-added

Figure 204: Create New Maid

Maid Code

Secret

Table 103: Create New Maid

Enter the Code to use when the maid wants to use the Mini Bar.

Enter the password associated with the maid.

In order to create a new consumer goods, click on

GUIValue-added FeaturesPMSMini Bar, the following window will popup.

under UCM web

Code

Name

Success Prompt

Failure Prompt

Figure 205: Create New Consumer Goods

Enter the Goods Code.

Enter the Name of the Goods

Select the success prompt when typing the code of the goods by the maid.

Select the failure prompt.

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The Minibar page displays as the following figure:

Figure 206: Mini Bar

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CRM INTEGRATION

Customer relationship management (CRM) is a term that refers to practices, strategies and technologies that companies use to manage and analyze customer interactions and data throughout the customer lifecycle, with the goal of improving business relationships with customers.

The UCM6100 series supports two CRM API, SugarCRM and Salesforce CRM, which allows users to look for contact information in the Contacts, Leads and / or Accounts tables, shows the contact record in CRM page, and saves the call information in the contact’s history.

SugarCRM

Configuration page of the SugarCRM can be accessed via admin login, on the UCM webGUIValue-

added FeaturesCRM.

Figure 207: SugarCRM Basic Settings

1.

Select “SugarCRM” from the CRM System Dropdown in order to use SugarCRM.

The following table gives details and descriptions of the parameters to configure SugarCRM integration with the UCM.

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CRM System

CRM Server Address

Add Unknown Number

Contact Lookups

Table 104: SugarCRM Settings

Select a CRM system from the Drop down, two CRM systems are available: Salesforce and SugarCRM.

Enter the IP address of the CRM server.

Add the new number to this module if it can’t be found in the selected module.

Select from the “Available” list of lookups and press to select where the UCM can perform the lookups on the CRM tables,

Leads, Accounts, and Contacts.

Once settings on admin access are configured:

2.

Click on and .

3.

Logout from admin access.

4.

Login to the UCM as user and navigate under “User PortalValue-added FeatureCRM User

Settings”.

Click on “Enable CRM” and enter the username/password associated with the CRM account then click on

and . The status will change from “Logged Out” to “Logged In”.

User can start then using SugarCRM features.

Figure 208: CRM User Settings

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Salesforce CRM

Configuration page of the Salesforce CRM can be accessed via admin login, on the UCM webGUI under

Value-added FeaturesCRM”.

Figure 209: Salesforce Basic Settings

1.

Select “Salesforce” from the CRM System Dropdown in order to use Salesforce CRM.

CRM System

Add Unknown Number

Table 105: Salesforce Settings

Select a CRM system from the Drop down, two CRM systems are available: Salesforce and SugarCRM.

Add the new number to this module if it can’t be found in the selected module.

Contact Lookups

Select from the “Available” list of lookups and press to select where the UCM can perform the lookups on the CRM tables, Leads, Accounts, and Contacts.

Once settings on admin access are configured:

2.

Click on and .

3.

Logout from admin access.

4.

Login to the UCM as user and navigate under “User PortalValue-added FeatureCRM User

Settings”.

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Click on “Enable CRM” and enter the username, password and Security Token associated with the CRM account then click on and . The status will change from “Logged Out” to “Logged

In”. User can start then using Salesforce CRM features.

Figure 210: Salesforce User Settings

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WAKEUP SERVICE

The WakeUp service can be used to schedule a reminder or wake up calls to any valid destination. This service is available on the UCM6100 as a separated module.

There are three ways to set up Wakeup Service:

• Using admin login

• Using user portal

• Using feature code

WakeUp Service using Admin Login

1.

Login to the UCM as admin.

2.

WakeUp service can be found under web GUIValue-added FeaturesWakeup Service”, click on to create a new wakeup service. The following window will popup.

3.

Click and

Figure 211: Create New Wakeup Service - Admin Login

to apply the changes.

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Enable Wakeup Service

Name

Prompt

Custom Date

Date

Time

Members

Table 106: Wakeup Service

Enable Wakeup service.

Enter a name to identify the wakeup service.

Select the prompt to play for that extension.

If disabled, users can select a specific date and time.

If enabled users can select multiple days of the week to perform the wakeup.

Select the date or dates when to performs the wakeup call.

Select the time when to play the wakeup call.

Select the members involved within the wakeup service group.

WakeUp Service from User portal

1.

Login to the user portal on the UCM6100.

2.

WakeUp service can be found under “Value-added FeaturesWakeup Service”, click on “Create

New Wakeup Service” to create a new wakeup service.

3.

Configures the Name, Prompt, Date and Time for the user to make the wakeup to.

4.

Click and to apply the changes.

WakeUp Service using Feature Code

1.

Login to the UCM as admin.

2.

Enable “Wakeup Service” from the WebGUI under “PBXCall FeaturesFeature Codes”.

Figure 212: Wakeup Service Feature Code

3.

Click “save” and “apply changes” to apply the changes.

4.

Dial “*36” which is the feature code by default to access to the UCM wakeup service in order to add, update, activate or deactivate UCM wakeup service.

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ANNOUNCEMENTS CENTER

The UCM6100 supports Announcements Center feature which allows users to pre-record and store voice message into UCM6100 with a specified code. The users can also create group with specified extensions.

When the code and the group number are dialed together in the combination of code + group number, the specified voice message is sent to all group members and only extensions in the group will hear the voice message.

Figure 213: Announcements Center

Announcements Center Settings

Name

Code

Table 107: Announcements Center Settings

Configure a name for the newly created Announcements Center to identify this announcement center.

Enter a code number for the custom prompt. This code will be used in combination with the group number. For example, if the code is 55, and group number is 666. The user can dial 55666 to send prompt 55 to all members in group 666.

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Custom Prompt

Note: The combination number must not conflict with any number in the system such as extension number or conference number.

This option is to set a custom prompt as an announcement to notify group members. The file can be uploaded from page ‘Custom Prompt’. Click

‘Prompt’ to add additional record.

Configure the ring timeout for the group members. Default value is 30sec.

Ring Timeout

Group Settings

Name

Number

Table 108: Group Settings

Configure a name for the newly created group to identify the group.

Configure the group number. The group number is used in combination with the code. For example, if group number is 666, and code is 55. The user can dial 55666 to send prompt 55 to all members in group 666.

Note:

The combination number must not conflict with any number in the system such as extension number or conference number.

Announcements Center feature can be found under web GUIValue-added FeaturesAnnouncements

Center. The following example demonstrates the usage of this feature.

1.

Click to create new group.

2.

Give a name to the newly created group.

3.

Create a group number which is used with code to send voice message.

4.

Select the extensions to be included in the group, who will receive the voice message.

Figure 214: Announcements Center Group Configuration

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In this example, group “Test” has number 666. Extension 1000, 1001 and 1002 are in this group.

1.

Click to create a new Announcement Center.

2.

Give a name to the newly created Announcement Center.

3.

Specify the code which will be used with group number to send the voice message to.

4.

Select the message that will be used by the code from the Custom Prompt drop down menu. To create a new Prompt, please click “Prompt” link and follow the instructions in that page.

Figure 215: Announcements Center Code Configuration

Code and Group number are used together to direct specified message to the target group. All extensions in the group will receive the message. For example, we can send code 55 to group 666 by dialing 55666 from any extension registered to the UCM6100. All the members in group 666 which are extension 1000,

1001 and 1002 will receive this voice message after they pick up the call.

Figure 216: Announcements Center Example

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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System

StatusDashboard.

Figure 217: StatusPBX Status

Trunks

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

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Figure 218: Trunk Status

Table 109: Trunk Status

Status

Trunks

Type

Display trunk status.

• Analog trunk status:

Available

Busy

Unavailable

Unknown Error

• SIP Peer trunk status:

Unreachable: The hostname cannot be reached.

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

Reachable: The hostname can be reached.

• SIP Register trunk status:

Registered

Unrecognized Trunk

Display trunk name

Display trunk Type:

• Analog

• SIP

• IAX

Username

Display username for this trunk.

Port/Hostname/IP

Display Port for analog trunk, or Hostname/IP for VoIP (SIP/IAX) trunk.

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Extensions

Extensions Status can be seen from the same configuration page, users can go under Web

GUIExtension/TrunkExtensions and following page will be displayed listing the extensions and their status information.

Figure 219: Extension Status

Status

Presence Status

Extension

CallerID Name

IP and Port

Email

Terminal Type

Table 110: Extension Status

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

• following definitions.

Green: Free

Blue: Ringing

Yellow: In Use

Grey: Unavailable

Display the presence status of the extension.

Display the extension number.

First name and last name of the extension.

Display the IP and port number of the registered device.

Display Email Notification status for the extension.

When notification is waiting for be sent, shows and once sent it will display

Displays extension type.

• SIP User

• IAX User

• Analog User

• Ring Groups

• Voicemail Groups

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Interfaces Status

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

Figure 220: UCM6102 Interfaces Status

Table 111: Interface Status Indicators

USB connected.

USB disconnected.

SD Card connected.

SD Card disconnected.

LAN/WAN connected.

LAN/WAN not configured.

LAN/WAN disconnected.

FXS/FXO connected.

FXS/FXO waiting.

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FXS/FXO busy.

FXS/FXO not configured.

FXS/FXO disconnected.

System Status

The UCM6100 system status can be accessed via Web GUIStatusSystem Status, which displays the following system information.

General

Network

Storage Usage

Resource Usage

General

Under Web GUISystem StatusSystem InformationGeneral, users could check the hardware and software information for the UCM6100. Please see details in the following table.

Table 112: System StatusGeneral

System Status System InformationGeneral

Model

Product model.

Part Number

System Time

Product part number.

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

Up Time

Boot

Core

Base

Program

Recovery

System up time since the last reboot.

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System StatusSystem InformationNetwork, users could check the network information for the UCM6100. Please see details in the following table.

Table 113: System StatusNetwork

System StatusSystem InformationNetwork

MAC Address

Global unique ID of device, in HEX format. The MAC address can be found on the label coming with original box and on the label located on the bottom of the device.

IP Address

IP address.

Gateway

Subnet Mask

DNS Server

Default gateway address.

Subnet mask address.

DNS Server address.

Storage Usage

Users could access the storage usage information from web GUISystem StatusDashboardStorage

Usage. It shows the available and used space for Space Usage and Inode Usage.

Space Usage includes:

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.

Inode Usage includes:

Configuration partition

Data partition

Note: Inode is the pointer used for file reference in the system. The system usually has limited resources of pointers.

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Figure 221: System StatusStorage 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System StatusDashboard

Resource Usage, the current CPU usage and Memory usage are shown in the pie chart.

Figure 222: System StatusResource 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MaintenanceSystem Events. The following event are currently supported on the UCM6100 which will have alert and/or Email generated if occurred:

Disk Usage

External Disk Usage

Modify Admin Password

Memory Usage

System Reboot

System Update

System Crash

Register SIP Failed

Register SIP Trunk Failed

Restore Config

User Login Success

User Login Failed

SIP Internal Call Failure

SIP Outgoing Call through Trunk Failure

Fail2ban Blocking

SIP Lost Registration

SIP Peer Trunk Status

Click on to configure the parameters for each event. See examples below.

1.

Disk Usage.

Figure 223: System EventsAlert 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.

External Disk Usage

Figure 224: System EventsAlert Events Lists: External Disk Usage

Detect Cycle: The UCM6100 will perform the External disk usage detection based on this cycle.

Users can enter the number and then select second(s)/minute(s)/hour(s)/day(s) to configure the cycle.

Alert Threshold: If the detected value exceeds the threshold (in percentage), the UCM6100 system will send the alert.

3.

Memory Usage

Figure 225: System EventsAlert 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.

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4.

System Crash

Figure 226: System EventsAlert Events Lists: System Crash

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

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

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

Alert Log

Under Web GUIMaintenanceSystem EventsAlert Log, system messages from triggered system events are listed as alert logs. The following screenshot shows system crash alert logs.

Figure 227: System EventsAlert Log

User could also filter alert logs by selecting a certain event category, type of alert log, and/or specifying a certain time period. The matching results will be displayed after clicking on classified into two types by the system:

. Alert logs are

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1.

Generate Alert: Generated when alert events happen, for example, alert logs for disk usage exceeding the alert threshold.

2.

Restore to Normal: Generated when alert events being cleared, for example, logs for disk usage dropping back below the alert threshold.

User could filter out alert logs of “Generate Alert” or “Restore to Normal” by specifying the type according to need. The following figure shows an example of filtering out alert logs of type of “Restore to Normal”.

Figure 228: Filter for Alert Log

Alert Contact

Users could add administrator's Email address under Web GUIMaintenanceSystem EventsAlert

Contact to send the alert notification to. Up to 10 Email addresses can be added.

CDR

CDR (Call Detail Record) is a data record generated by the PBX that contains attributes specific to a single instance of phone call handled by the PBX. It has several data fields to provide detailed description for the call, such as phone number of the calling party, phone number of the receiving party, start time, call duration, and etc.

On the UCM6100, the CDR can be accessed under Web GUICDRCDR. Users could filter the call report by specifying the date range and criteria, depending on how the users would like to include the logs to the report. Click on "Search" button to display the generated report.

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Figure 229: CDR Filter

Call Type

Table 114: CDR Filter Criteria

Groups the following:

Inbound calls: Inbound calls are calls originated from a non-internal source

(like a VoIP trunk) and sent to an internal extension.

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

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

External calls: External calls are calls sent from one trunk to another trunk, which are not sent to any internal extension.

Status

Filter with the call status, the available statuses are the following:

• Answered

• No Answer

• Busy

• Failed

Source Trunk

Name

Destination Trunk

Name

Select source trunk(s) and the CDR of calls going through inbound the trunk(s) will be filtered out.

Select destination trunk(s) and the CDR of calls going outbound through the trunk(s) will be filtered out.

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Action Type

Account Code

Start Time

End Time

Caller Number

Filter calls using the Action Type, the following actions are available:

• Dial

• Announcements

• Callback

• Call Forward

• Conference

• Disa

• Fax

• Follow Me

• IVR

• Page

• Parked Call

• Queue

• Ring Group

• Transfer

• VFax

• VM

• VMG

• Wakeup

Select the account Code to filter with. If pin group CDR is enabled, the call with pin group information will be displayed as part of the CDR under Account Code Field.

Specify the start time to filter the CDR report. Click on the calendar icon on the right and the calendar will show for users to select the exact date and time.

Specify the end time to filter the CDR report. Click on the calendar icon on the right and the calendar will show for users to select the exact date and time.

Enter the caller number to filter the CDR report. CDR with the matching caller number will be filtered out.

User could specify a particular caller number or enter a pattern. ‘.’ matches zero or more characters, only appears in the end. ‘X’ matches any digit from 0 to 9, caseinsensitive, repeatable, only appears in the end.

Caller Name

Callee Number

For example:

3XXX: It will filter out CDR that having caller number with leading digit 3 and of 4 digits’ length.

3.: It will filter out CDR that having caller number with leading digit 3 and of any length.

Enter the caller name to filter the CDR report. CDR with the matching caller name will be filtered out.

Enter the callee number to filter the CDR report. CDR with the matching callee number will be filtered out.

The call report will display as the following figure shows.

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Figure 230: Call Report

The CDR report has the following data fields:

Start Time

Format: 2016-09-03 00:06:16

Call Type

Example:

IVR

DIAL

WAKEUP

Call From

Example format:

"John Doe" 2000

Call To

Example format:

2002

Call Time

Format: 0:00:02

Talk Time

Format: 0:00:00

Account Code

Example format:

Grandstream/Test

Status

Answered, Busy, No answer or Failed.

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Users could perform the following operations on the call report.

Sort by “Start Time”

Click on the header of the column to sort the report by "Start Time". Clicking on "Start Time" again will reverse the order.

Download Searched Results

Click on “Download Search Result(s)” to export the records filtered out to a .csv file.

Download All Records

Click on “Download All Records” to export all the records to a .csv file.

Delete All

On the bottom of the page, click on "Delete All" button to remove all the call report information.

Play/Download/Delete Recording File (per entry)

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 231: Call Report Entry with Audio Recording File

Automatic Download CDR Records

User could configure the UCM6100 to automatically download the CDR records and send the records to multiple email recipients in a specific hour. Click on “Automatic Download Settings”, and configure the parameters in the dialog below.

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Figure 232: Automatic Download Settings

To receive CDR record automatically from Email, check “Enable” and select a time period “By Day” “By

Week” or “By Month”, select Hour of the day as well for the automatic download period. Make sure you have entered an Email or multiple email addresses where to receive the CDR records.

CDR Improvement

Starting from UCM6100 firmware 1.0.10.x, transferred call will no longer be displayed as a separate call entry in CDR. It will display within call record in the same entry. CDR new features can be found under web

GUICDRCDR. The user can click on the option icon for a specific call log entry to view details about this entry, such as premier caller and transferred call information.

Figure 233: CDR Report

Figure 234: Detailed CDR Information

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Downloaded CDR File

The downloaded CDR (.csv file) has different format from the Web GUI CDR. Here are some descriptions.

Caller number, Callee number

"Caller number": the caller ID.

"Callee number": the callee ID.

If the "Source Channel" contains "DAHDI", this means the call is from FXO/PSTN line.

Figure 235: Downloaded CDR File Sample

Context

There are different context values that might show up in the downloaded CDR file. The actual value can vary case by case. Here are some sample values and their descriptions.

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 236: 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.

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Sample 2:

Figure 237: Downloaded CDR File Sample - Source Channel and Dest Channel 2

"PJSIP" means it's a SIP call. There are three possible format:

(a) PJSIP/NUM-XXXXXX, where NUM is the local SIP extension number. The last XXXXX is a random string and can be ignored.

(c) PJSIP/trunk_X/NUM, where trunk_X is the internal trunk name, and NUM is the number to dial out through the trunk.

(c) PJSIP/trunk_X-XXXXXX, where trunk_X is the internal trunk name and it is an inbound call from this trunk. The last XXXXX is a random string and can be ignored.

There are some other possible values, but these values are almost the application name which are used by the dialplan.

IAX2/NUM-XXXXXXX: it means this is an IAX call.

Local/@from-internal-XXXXX: it is used internally to do some special feature procedure. We can simply ignore it.

Hangup: the call is hung up from the dialplan. This indicates there are some errors or it has run into abnormal cases.

Playback: play some prompts to you, such as 183 response or run into an IVR.

ReadExten: collect numbers from user. It may occur when you input PIN codes or run into DISA

Statistics

CDR Statistics is an additional feature on the UCM6100 which provides users a visual overview of the call report across the time frame. Users can filter with different criteria to generate the statistics chart.

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Trunk Type

Call Type

Time Range

Figure 238: CDR Statistics

Table 115: 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.

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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 239: CDRRecording Files

• Click on “Delete Selected Recording Files” to delete the recording files.

• Click on “Delete All Recording Files” to delete all recording files.

• Click on

to download the recording file in .wav format.

• Click on

to delete the recording file.

• To sort the recording file, click on the title "Caller", "Callee" or "Call Time" for the corresponding column.

Click on the title again can switch the sorting mode between ascending order or descending order.

API Configuration

The UCM6100 supports third party billing interface API for external billing software to access CDR and call recordings on the PBX. The API uses HTTPS to request the CDR data and call recording data matching given parameters as configured on the third-party application.

Before accessing the API, the administrators need enable API and configure the access/authentication information on the UCM6100 first. The API configuration parameters are listed in the table below.

Table 116: API Configuration Files

Enable

Enable/Disable API. The default setting is disabled.

TLS Bind Address Configure the IP address for TLS server to bind to. "0.0.0.0" means binding to all interfaces. The port number is optional and the default port number is 8443. The IP address must match the common name (host name) in the certificate so that the TLS socket won't bind to multiple IP addresses. The default setting is 0.0.0.0:8443.

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Username

Password

Permitted IPs

TLS Private Key

TLS Cert

Configure the Username for API Authentication.

Configure the Password for API Authentication.

Specify a list of IP addresses permitted by API. This creates an IP-specific access control list. Multiple entries are allowed.

For example, "192.168.5.20/255.255.255.255" denies access from all IP addresses except 192.168.5.20

The default setting is blank, meaning all IPs will be denied. Users must set permitted

IP address before connecting to the API.

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.

Reset Certificates

Press button to restore UCM’s default certificates.

For more details on CDR API (Access to Call Detail Records) and REC API (Access to Call Recording

Files), please refer the document in the link here: http://www.grandstream.com/sites/default/files/Resources/ucm6xxx_cdr_rec_api_guide.pdf

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USER PORTAL

Users could log into their web GUI portal using the extension number and user password. When an extension is created in the UCM61XX, the corresponding user account for the extension is automatically created. The user portal allows access to a variety of features which include user information, extension configuration and CDR as well as settings and managing value-added features like webRTC, Fax Sending,

Call Queue, Wakeup Service and CRM.

Users also can access their personal data files (call recordings, Fax files, Voicemail Prompts …).

The login credentials are configured by Super Admin. The following figure shows the dialog of editing the account information by Super Admin. The User Name must be the extension number and it’s not configurable, and the password is set on “User Password” field and it should not be confused with the SIP extension password.

Figure 240: Edit User Information by Super Admin

The following screenshot shows an example of login page using extension number 1000 as the username.

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Figure 241: User Portal Login

After login, the Web GUI display is shown as below.

Figure 242: User Portal Layout

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After successful login, the user has the following three configuration tabs:

Basic Information

Under this menu, the user can configure and change his/her personal information including (first name, last name, password, email address, department…). And they can also set and activate their extension features

(presence status, call forward, DND ….) to be reflected on the UCM.

Also, the user can see from this menu the Call Details Records and search for specific ones along with the possibility to download the records on CSV format for later usage.

Personal Data

Under this section, the user can access and manage their personal data files which includes (voicemail files, call recordings, and fax files) along with the possibility to set Follow me feature to without requesting the Super admin to set the feature from admin account.

Value-added Features

On this section, the user has access to manage and use all rich value-added features which includes.

+ webRTC connection and making calls from the browser.

+ Sending Fax files using PDF or TIF/TIFF format.

+ If user is a member of call queue, they can check the queue’s activity from the “Call Queue” section.

+ Create and enable WakeUp service.

+ Enable and configure CRM connection to either SugarCRM or Salesforce.

For the configuration parameter information in each page, please refer to

[Table 117: User

Management

Create New User]

for options in User PortalBasic InformationPersonal Config page; please refer to

[EXTENSIONS]

for options in User PortalBasic InformationExtensions page; please refer to

[CDR]

for User PortalBasic InformationCDR page.

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MAINTENANCE

User Management

User management is on web GUIMaintenanceUser Management page. User could create multiple accounts for different administrators to log in the UCM6100 web GUI. Additionally, the system will automatically create user accounts along with creating new extensions for extension users to login to the

Web GUI using their extension number and password. All existing user accounts for Web GUI login will be displayed on User Management page as shown in the following figure.

Figure 243: User Management Page Display

User Information

When logged in as Super Admin, click on to create a new account for Web GUI user. The following dialog will prompt. Configure the parameters as shown in below table.

Figure 244: Create New User

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User Name

User Password

Table 117: User ManagementCreate New User

Configure a username to identify the user which will be required in Web GUI login.

Letters, digits and underscore are allowed in the user name.

Configure a password for this user which will be required in Web GUI login. Letters, digits and underscore are allowed.

This is the role of the Web GUI user. Currently “Administrator” and “Custom

Privilege” are supported when Super Admin creates a new user.

Privilege

Department

Fax

Email Address

First Name

Last Name

Home Number

Phone Number

Enter the necessary information to keep a record for this user.

Once created, the Super Admin can edit the users by clicking on or delete the user by clicking on

Figure 245: User Management – New Users

Custom Privilege

Four privilege levels are supported:

Super Administrator

-

This is the highest privilege. Super Admin can access all pages on UCM6100 Web GUI, change configuration for all options and execute all the operations.

-

Super Admin can create, edit and delete one or more users with “Admin” privilege

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Super Admin can edit and delete one or more users with “Consumer” privilege

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Super Admin can view operation logs generated by all users.

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By default, the user account “admin” is configured with “Super Admin” privilege and it’s the only user with “Super Admin” privilege. The User Name and Privilege level cannot be changed/deleted.

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-

Super Admin could change its own login password on Web GUIMaintenanceChange

Information page.

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Super Admin could view operations done by all the users in Web GUIMaintenanceUser

ManagementOperation Log

Administrator

-

Users with “Admin” privilege can only be created by “Super Admin” user.

-

“Admin” privilege users are not allowed to access the following pages:

MaintenanceUpgrade

MaintenanceBackup

MaintenanceCleaner

MaintenanceReset/Reboot

SettingsUser ManagementOperation Log

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“Admin” privilege users cannot create new users for login.

Consumer Privilege

-

A user account for Web GUI login is created automatically by the system when a new extension is created.

-

The user could log in the Web GUI with the extension number and password to access user information, extension configuration and CDR of that extension.

-

The SuperAdmin user can click on the “General_User” in order to enable/disable the custom privilege from deleting their own recording files, changing SIP credentials and disabling voicemail service in their user portal account.

Figure 246: General User

Custom Privilege

The Super Admin user can create users with different privileges. 6 modules are available for privilege customization.

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System Status

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Conference

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System Events

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CDR Records

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CDR API

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Wakeup Service

Figure 247: Create New Custom Privilege

Log in UCM6100 as super admin and go to MaintenanceUser ManagementCustom Privilege, create privilege with customized available modules.

To assign custom privilege to a sub-admin, navigate to UCM webUIMaintenanceUser

ManagementUser InformationCreate New User/Edit Users, select the custom privilege from

“Privilege” option.

Concurrent Multi-User Login

When there are multiple Web GUI users created, concurrent multi-user login is supported on the UCM6100.

Multiple users could edit options and have configurations take effect simultaneously. However, if different users are editing the same option or making the same operation (by clicking on “Apply Changes”), a prompt will pop up as shown in the following figure.

Figure 248: Multiple User Operation Error Prompt

Operation Log

Super Admin has the authority to view operation logs on UCM6100 web GUIMaintenanceUser

ManagementOperation Log page. Operation logs list operations done by all the Web GUI users, for example, Web GUI login, creating trunk, creating outbound rule and etc.

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There are 7 columns to record the operation details “Date”, “User Name”, “IP Address”, “Results”, “Page

Operation”, “Specific Operation” and “Remarks”.

Figure 249: Operation Logs

The operation log can be sorted and filtered for easy access. Click on the header of each column to sort.

For example, clicking on "Date" will sort the logs according to operation date and time. Clicking on "Date" again will reverse the order.

Date

User Name

IP Address

Results

Page Operation

Table 118: Operation Log Column Header

The date and time when the operation is executed.

The username of the user who performed the operation.

The IP address from which the operation is made.

The result of the operation.

The page where the operation is made. For example, login, logout, delete user, create trunk and etc.

Specific Operation

Remark

Click on to view the options and values configured by this operation.

Allows users to add notes and remarks to each operation

User could also filter the operation logs by time condition, IP address and/or username. Configure these conditions and then click on .

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Figure 250: Operation Logs Filter

The above figure shows an example that operations made by user “support” on device with IP

192.168.40.173 from 2014-11-01 00:00 to 2014-11-06 15:38 are filtered out and displayed.

To delete operation logs, users can perform filtering first and then click on to delete the filtered result of operation logs. Or users can click on to delete all operation logs at once.

Change Password

After logging in the UCM6100 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MaintenanceUser ManagementChange Information page.

2.

Enter the old password first.

3.

Enter the new password and re-type the new password to confirm. The new password has to be at least 4 characters. The maximum length of the password is 30 characters.

4.

Configure the Email Address that is used when login credential is lost.

5.

Click on "Save" and the user will be automatically logged out.

6.

Once the web page comes back to the login page again, enter the username "admin" and the new password to login.

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Figure 251: Change Password

Enter Old Password

Enter New Password

Retype New Password

Email Address

Enter the Old Password for UCM6100

Enter the New Password for UCM6100

Retype the New Password for UCM6100

Configure the Email address for UCM6100. In case login credential is lost,

Email address is used to retrieve login credential

Change Binding Email

UCM6100 allows user to configure binding email in case login password is lost. UCM6100 login credentials will be sent to the designated email address. The feature can be found under Web

GUIMaintenanceUser ManagementChange Information.

Figure 252: Change Binding Email

Table 119: Change Binding Email option

Enter the password of the account

Email Address

Enter the current login user credential for UCM6100

Email Address is used to retrieve password when password is lost

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Login Settings

After the user logs in the UCM6100 Web GUI, the user will be automatically logged out after certain timeout, or he/she can be banned for a specific period if the login timeout is exceeded. Those values can be specified under UCM6100 web GUIMaintenanceChange InformationLogin Settings page.

The “User Login Timeout” value is in minute and the default setting is 10 minutes. If the user doesn’t make any operation on Web GUI within the timeout, the user will be logged out automatically. After that, the Web

GUI will be redirected to the login page and the user will need to enter username and password to log in.

If set to 0, there is no timeout for the Web GUI login session and the user will not be automatically logged out.

Maximum number of login attempts” can prevent the UCM6100 from brutal force decryption, if this number is exceeded user IP address will be banned from accessing the UCM for a period of time based on user configuration, the default value is 5.

User ban period” specify the period of time in minutes an IP will be banned from accessing the UCM if the User max number of try login is exceeded, the default value is 5.

Login Banned User List show the list of IPs’ banned from the UCM.

Login White List” User can add a list of IPs’ to avoid the above restriction, thus, they can exceed the User max number of try login.

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Figure 253: Login Timeout Settings

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/BETA

The upgrading configuration can be accessed via Web GUIMaintenanceUpgrade.

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Figure 254: Network Upgrade

Table 120: Network Upgrade Configuration

Upgrade Via

Allow users to choose the firmware upgrade method: TFTP, HTTP or

HTTPS.

Firmware Server Path

Firmware File Prefix

Define the server path for the firmware server.

If configured, only the firmware with the matching encrypted prefix will be downloaded and flashed into the UCM6100.

Firmware File Suffix

If configured, only the firmware with the matching encrypted postfix will be downloaded and flashed into the UCM6100.

HTTP/HTTPS User Name

HTTP/HTTPS Password

The user name for the HTTP/HTTPS server.

The password for the HTTP/HTTPS server.

Please follow the steps below to upgrade the firmware remotely.

• Enter the firmware server path under Web GUIMaintenanceUpgrade.

• 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.

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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.

1.

Download the latest UCM6100 firmware file from the following link and save it in your PC.

2. http://www.grandstream.com/support/firmware

3.

Log in the Web GUI as administrator in the PC.

4.

Go to Web GUIMaintenanceUpgrade, upload the firmware file by clicking on “choose file to upload” and select the firmware file from your PC. The default firmware file name is ucm6100fw.bin

5.

Click on

Figure 255: Local Upgrade

to start upgrading.

Figure 256: Upgrading Firmware Files

6.

Wait until the upgrading process is successful and a window will be popped up in the Web GUI.

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Figure 257: Reboot UCM6100

7.

Click on "OK" to reboot the UCM6100 and check the firmware version after it boots up.

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Notes:

• Please do not interrupt or power cycle the UCM6100 during upgrading process.

• The firmware file name allows the use of the following special characters: “[email protected]#*~&”.

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No Local Firmware Servers

Service providers should maintain their own firmware upgrade servers. For users who do not have

TFTP/HTTP/HTTPS server, some free windows version TFTP servers are available for download from http://www.solarwinds.com/products/freetools/free_tftp_server.aspx

http://tftpd32.jounin.net

Please check our website at http://www.grandstream.com/support/firmware for latest firmware.

Instructions for local firmware upgrade via TFTP:

1.

Unzip the firmware files and put all of them in the root directory of the TFTP server;

2.

Connect the PC running the TFTP server and the 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.

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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.

Backup/Restore

Users could backup the UCM6100 configurations for restore purpose under Web

GUIMaintenanceBackupBackup/Restore.

Click on to create a new backup file. Then the following dialog will show.

Figure 258: Create New Backup

1.

Choose the type(s) of files to be included in the backup.

2.

Choose where to store the backup file: USB Disk, SD Card or Local.

3.

Name the backup file.

4.

Click on "Backup" to start backup.

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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 UCM61xx internal storage or the external device.

Click on to upload backup file from the local device to UCM61xx. The uploaded backup file will also be displayed in the web page and can be used to restore the UCM61xx.

It is also possible to load backup files from UCM62xx to UCM61xx and vice versa.

Please make sure the FXO port settings, total number of extensions and total number of conference rooms are compactable before restoring to another UCM model. Otherwise it will prompt a warning and stop the restore process as shown below:

Figure 259: Restore Warning

Figure 260: Backup / Restore

The option allows UCM to perform automatically backup on the user specified time.

Regular backup file can only be stored in USB / SD card / SFTP server. User is allowed to set backup time from 0-23 and how frequent the backup will be performed.

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Figure 261: 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MaintenanceBackupData Sync.

The client account supports special characters such as @ or “.”. This change allows user to use email address as SFTP accounts. It allows users as well to specify the destination directory on SFTP server for backup file. If the directory doesn’t exist on the destination, UCM6100 will create the directory automatically.

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Enable Data Sync

Account

Password

Server Address

Destination Directory

Sync Time

Figure 262: Data Sync

Table 121: Data Sync Configuration

Enable the auto data sync 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.

Enter the SFTP server address.

Specify the directory in SFTP server to keep the backup file. Format:

‘xxx/xxx/xxx’, If this directory does not exist, UCM will create this directory automatically.

Enter 0-23 to specify the backup hour of the day.

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Before saving the configuration, users could click on "Test Connection". The UCM6100 will then try connecting the server to make sure the server is up and accessible for the UCM6100. Save the changes and all the backup logs will be listed on the web page. After data sync is configured, users could also

instead of waiting for the backup time manually synchronize all data by clicking on interval to come.

Restore Configuration from Backup File

To restore the configuration on the UCM6100 from a backup file, users could go to Web

GUIMaintenanceBackupBackup/Restore.

• A list of previous configuration backups is displayed on the web page. Users could click on

of the desired backup file and it will be restored to the UCM6100.

• If users have other backup files on PC to restore on the UCM6100, click on "Upload Backup File" first and select it from local PC to upload on the UCM6100. Once the uploading is done, this backup file will be displayed in the list of previous configuration backups for restore purpose. Click on from the backup file.

to restore

Figure 263: Restore UCM6100 from Backup File

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Note:

• The uploaded backup file must be a tar file with no special characters like *,!,#,@,&,$,%,^,(,),/,\,space in the file name.

• The uploaded back file size must be under 10MB.

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System Cleanup/Reset

Reset and Reboot

Users could perform reset and reboot under Web GUIMaintenanceSystem Cleanup/ResetReset

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, recording files, fax files, voice prompt, Music on Hold, CDR records, zero config files, operation log, backup files, core files, troubleshooting files, email files will be cleared.

All

All the configurations and data will be reset to factory default.

Figure 264: Reset and Reboot

Cleaner

Users could configure to clean the Call Detail Report/Voice Records/Voice Mails/FAX automatically under

Web GUIMaintenanceSystem Cleanup/ResetCleaner.

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Enable CDR Cleaner

Clean Files in External

Device

CDR Clean Time

Clean Interval

Enable Queue Log Cleaner

Queue Log Clean Time

Clean Interval

Enable File Cleaner

Figure 265: Cleaner

Table 122: Cleaner Configuration

Enable the CDR Cleaner function.

If enabled the files in external device (USB/SD card) will be atomically cleaned up as configured.

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.

Enable scheduled queue log cleaning. By default, is disabled.

Enter the hour of the day to start the cleaning. The valid range is 0-23.

Enter how often (in days) to clean queue logs. The valid range is 1-30.

If enabled, the files will be automatically cleaned up as configured.

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Choose Cleaner File

File Clean Threshold

File Clean Time

File Clean Interval

Select the files for system automatic clean.

• Basic Call Recording Files.

• Conference Recording Files.

• Call Queue Recording Files.

• Voicemail Files.

• Fax

• Backup Files.

Specify the threshold of local storage usage from 0 to 99 (in percentage).

Enter 0-23 to specify the hour of the day to clean up the files.

Enter 1-30 to specify the day of the month to clean up the files.

All the cleaner logs will be listed on the bottom of the page.

USB/SD Card Files Cleanup

Users are able to clean the Call Detail Report/Voice Records/Voice Mails/FAX automatically under Web

GUIMaintenanceCleanerUSB / SD Card Files Cleanup.

Figure 266: USB/SD Card Files Cleanup

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Current Path

Directory

Delete Selected File

Table 123: USB/SD Card Files Cleanup

Displays the current path.

Select the directory user want to clean.

Select multiple entries to delete from USB or SD card.

Syslog

On the UCM6100, users could dump the syslog information to a remote server under Web

GUIMaintenanceSyslog. 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 "notice" and "verb" levels besides "error" level. pbx: This module is related to general PBX functions. chan_sip: This module is related to SIP calls. chan_dahdi: This module is related to analog calls (FXO/FXS). app_meetme: This module is related to conference bridge.

----------------------------------------------------------------------------------------------- ---------------------------------------------

Note:

Syslog is usually for debugging and troubleshooting purpose. Turning on all levels for all syslog modules is not recommended for daily usage. Too many syslog print might cause traffic and affect system performance.

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Network Troubleshooting

On the UCM6100, users could capture traces, ping remote host and traceroute remote host for troubleshooting purpose under Web GUIMaintenanceNetwork Troubleshooting.

Ethernet Capture

The captured trace can be downloaded for analysis. The instructions or result will be displayed in the web

GUI output result.

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Figure 267: Ethernet Capture

Interface Type

Enable SFTP Data Sync

Storage to External Device

Capture Filter

Start

Stop

Download

Table 124: Ethernet Capture

Select the network interface to monitor.

Check this box to save the capture files in the SFTP server. Please make sure the configuration of data synchronization works before.

Check this box to activate storage of the capture either on the USB or SD Card.

Enter the filter to obtain the specific types of traffic, such as (host, src, dst, net, proto…).

Click to start the trace.

Click to stop the trace.

Click to download the trace if trace is stored locally.

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 268: 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 269: Traceroute

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Signaling Troubleshooting

Analog Record Trace

1.

Analog Record Trace

Analog record trace can be used to troubleshoot analog trunk issue, for example, the UCM6100 user has caller ID issue for incoming call from Analog trunk. Users can access analog record trance under web

GUIMaintenanceSignal TroubleshootingAnalog 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 270: Troubleshooting Analog Trunks

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2.

A key Dial-up FXO

Users can directly set a PSTN number on the “External Extension” text box to troubleshoot issues related to the analog trunk easily, the following steps shows how to use this feature:

1.

Configure analog trunk on UCM, including outbound route.

2.

Enter a reachable external number in “External Extension”.

3.

Press “Start” button. The call will be initiated to the external number.

4.

Answer and finish the call before pressing “Stop” button.

5.

The trace will be available for analysis to download after output result shows “Done! Click on

Download to download the captured packets”.

Figure 271: A Key Dial-up FXO

Note: When using a Key Dial-up FXO feature the outbound trunk for the analog trunk need to have internal permission. As well as it should be the trunk with the highest outbound route priority.

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 UCM6100. 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 UCM6100.

The default setting is 3. If there is no response from UCM6100 after 3 attempts (default) to check, current status will be stored and the internal service in UCM6100 will be restarted.

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Figure 272: Service Check

Network Status

In UCM6100 Web GUISystem StatusNetwork Status, the users can view active Internet connections.

This information can be used to troubleshoot connection issue between UCM6100 and other services.

Figure 273: Network Status

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EXPERIENCING THE UCM6100 SERIES IP PBX

Please visit our website: http://www.grandstream.com

to receive the most up- to-date updates on firmware releases, additional features, FAQs, documentation and news on new products.

We encourage you to browse our product related documentation , FAQs and User and Developer Forum for answers to your general questions. If you have purchased our products through a Grandstream Certified

Partner or Reseller, please contact them directly for immediate support.

Our technical support staff is trained and ready to answer all of your questions. Contact a technical support member or submit a trouble ticket online to receive in-depth support.

Thank you again for purchasing Grandstream UCM6100 series IP PBX appliance, it will be sure to bring convenience and color to both your business and personal life.

* Asterisk is a Registered Trademark of Digium, Inc.

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FCC Caution:

Any Changes or modifications not expressly approved by the party responsible for compliance could void the user's authority to operate the equipment.

This device complies with part 15 of the FCC Rules. Operation is subject to the following two conditions:

(1) This device may not cause harmful interference, and (2) this device must accept any interference received, including interference that may cause undesired operation.

Note: This equipment has been tested and found to comply with the limits for a Class B digital device, pursuant to part 15 of the FCC Rules. These limits are designed to provide reasonable protection against harmful interference in a residential installation. This equipment generates, uses and can radiate radio frequency energy and, if not installed and used in accordance with the instructions, may cause harmful interference to radio communications. However, there is no guarantee that interference will not occur in a particular installation. If this equipment does cause harmful interference to radio or television reception, which can be determined by turning the equipment off and on, the user is encouraged to try to correct the interference by one or more of the following measures:

- Reorient or relocate the receiving antenna.

- Increase the separation between the equipment and receiver.

- Connect the equipment into an outlet on a circuit different from that to which the receiver is connected.

- Consult the dealer or an experienced radio/TV technician for help.

Regulatory Information

U.S. FCC Part 68 Statement

This equipment complies with Part 68 of the FCC rules and the requirements adopted by the ACTA. The unit bears a label on the back which contains among other information a product identifier in the format US:

GNIIS00BUCM6104/US: GNIIS00BUCM6116. If requested, this number must be provided to the telephone company.

This equipment uses the following standard jack types for network connection: RJ11C.

This equipment contains an FCC compliant modular jack. It is designed to be connected to the telephone network or premises wiring using compatible modular plugs and cabling which comply with the requirements of FCC Part 68 rules.

The Ringer Equivalence Number, or REN, is used to determine the number of devices which may be connected to the telephone line. An excessive REN may cause the equipment to not ring in response to an incoming call. In most areas, the sum of the RENs of all equipment on a line should not exceed five (5.0).

In the unlikely event that this equipment causes harm to the telephone network, the telephone company can temporarily disconnect your service. The telephone company will try to warn you in advance of any such disconnection, but if advance notice isn't practical, it may disconnect the service first and notify you as soon as possible afterwards. In the event such a disconnection is deemed necessary, you will be advised of your right to file a complaint with the FCC.

From time to time, the telephone company may make changes in its facilities, equipment, or operations which could affect the operation of this equipment. If this occurs, the telephone company is required to provide you with advance notice so you can make the modifications necessary to obtain uninterrupted service.

There are no user serviceable components within this equipment. See Warranty flyer for repair or warranty information.

It shall be unlawful for any person within the United States to use a computer or other electronic device to send any message via a telephone facsimile unless such message clearly contains, in a margin at the top or bottom of each transmitted page or on the first page of the transmission, the date and time it is sent and an identification of the business, other entity, or individual sending the message and the telephone number of the sending machine or of such business, other entity, or individual. The telephone number provided may not be a 900 number or any other number for which charges exceed local or long-distance transmission charges. Telephone facsimile machines manufactured on and after December 20, 1992, must clearly mark such identifying information on each transmitted message. Facsimile modem boards manufactured on and after December 13, 1995, must comply with the requirements of this section.

This equipment cannot be used on public coin phone service provided by the telephone company.

Connection to Party Line Service is subject to state tariffs. Contact your state public utility commission, public service commission, or corporation commission for more information.

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