Grandstream Networks UCM6100 Series User manual

Grandstream Networks, Inc.

UCM6100 Series IP PBX

User Manual

UCM6100 Series IP PBX User Manual

Table of Content

GNU GPL INFORMATION ......................................................................... 16

CHANGE LOG ........................................................................................... 17

FIRMWARE VERSION 1.0.9.26 ......................................................................................................... 17




FIRMWARE VERSION 1.0.7.11 .......................................................................................................... 19

FIRMWARE VERSION 1.0.6.11 .......................................................................................................... 20




FIRMWARE VERSION 1.0.4.7 ........................................................................................................... 21




WELCOME ................................................................................................. 25

PRODUCT OVERVIEW ............................................................................. 27

FEATURE HIGHTLIGHTS .................................................................................................................. 27

TECHNICAL SPECIFICATIONS ......................................................................................................... 28

INSTALLATION .......................................................................................... 31

EQUIPMENT PACKAGING ................................................................................................................ 31

CONNECT YOUR UCM6100 .............................................................................................................. 31

CONNECT THE UCM6102 .......................................................................................................... 31

CONNECT THE UCM6104 .......................................................................................................... 32

CONNECT THE UCM6108 .......................................................................................................... 33

CONNECT THE UCM6116 .......................................................................................................... 34

SAFETY COMPLIANCES ................................................................................................................... 35

WARRANTY ....................................................................................................................................... 35

GETTING STARTED .................................................................................. 37

USE THE LCD MENU ......................................................................................................................... 37

USE THE LED INDICATORS .............................................................................................................. 39

Firmware Version 1.0.9.26 UCM6100 Series IP PBX User Manual Page 1 of 304

USE THE WEB GUI ............................................................................................................................ 40

ACCESS WEB GUI ...................................................................................................................... 40

WEB GUI CONFIGURATIONS .................................................................................................... 41

WEB GUI LANGUAGES .............................................................................................................. 42

SAVE AND APPLY CHANGES .................................................................................................... 42

MAKE YOUR FIRST CALL ................................................................................................................. 42

SYSTEM SETTINGS .................................................................................. 45

USER MANAGEMENT ....................................................................................................................... 45

USER PRIVILEGES ..................................................................................................................... 45

CREATE NEW WEB UI USER ..................................................................................................... 46

USER PORTAL ............................................................................................................................ 48

CONCURRENT MULTI-USER LOGIN ......................................................................................... 49

OPERATION LOG ........................................................................................................................ 49

CHANGE BINDNG EMAIL ........................................................................................................... 51

NETWORK SETTINGS ....................................................................................................................... 52

BASIC SETTINGS ....................................................................................................................... 52

802.1X .......................................................................................................................................... 59

STATIC ROUTES ......................................................................................................................... 60

PORT FORWORDING (UCM6102 ONLY) ................................................................................... 63

DDNS SETTINGS ........................................................................................................................ 65

FIREWALL .......................................................................................................................................... 66

STATIC DEFENSE ....................................................................................................................... 66

DYNAMIC DEFENSE ................................................................................................................... 69

FAIL2BAN .................................................................................................................................... 70

CHANGE PASSWORD ....................................................................................................................... 71

LDAP SERVER ................................................................................................................................... 71

LDAP SERVER CONFIGURATIONS ........................................................................................... 72

LDAP PHONEBOOK .................................................................................................................... 73

LDAP CLIENT CONFIGURATIONS ............................................................................................. 77

HTTP SERVER ................................................................................................................................... 79

EMAIL SETTINGS .............................................................................................................................. 79

TIME SETTINGS ................................................................................................................................ 80

AUTO TIME UPDATING .............................................................................................................. 80

SET TIME MANUALLY ................................................................................................................ 82

OFFICE TIME .............................................................................................................................. 83

HOLIDAY ...................................................................................................................................... 84

NTP SERVER ..................................................................................................................................... 86

RECORDINGS STORAGE ................................................................................................................. 87

LOGIN TIMEOUT SETTINGS ............................................................................................................. 89

PROVISIONING ......................................................................................... 91


OVERVIEW ......................................................................................................................................... 91

CONFIGURATION ARCHITECTURE FOR END POINT DEVICE ..................................................... 91

AUTO PROVISIONING SETTINGS .................................................................................................... 92

DISCOVERY ....................................................................................................................................... 95

GLOBAL CONFIGURATION ............................................................................................................... 96

GLOBAL POLICY ......................................................................................................................... 96

GLOBAL TEMPLATES ............................................................................................................... 103

MODEL CONFIGURATION .............................................................................................................. 106

MODEL TEMPLATES ................................................................................................................ 106

MODEL UPDATE ....................................................................................................................... 108

DEVICE CONFIGURATION .............................................................................................................. 109

CREATE NEW DEVICE ............................................................................................................. 109

MANAGE DEVICES ................................................................................................................... 110

SAMPLE APPLICATION ................................................................................................................... 116

EXTENSIONS .......................................................................................... 121

CREATE NEW USER ....................................................................................................................... 121

CREATE NEW SIP EXTENSION ............................................................................................... 121

CREATE NEW IAX EXTENSION ............................................................................................... 127

CREATE NEW FXS EXTENSION .............................................................................................. 131

BATCH ADD EXTENSIONS ............................................................................................................. 136

BATCH ADD SIP EXTENSIONS ................................................................................................ 136

BATCH ADD IAX EXTENSIONS ................................................................................................ 139

SEARCH AND EDIT EXTENSION .................................................................................................... 141

EXPORT EXTENSIONS ................................................................................................................... 143

IMPORT EXTENSIONS .................................................................................................................... 143

EMAIL TO USER .............................................................................................................................. 144

TRUNKS ................................................................................................... 147

ANALOG TRUNKS ........................................................................................................................... 147

ANALOG TRUNK CONFIGURATION ........................................................................................ 147

PSTN DETECTION .................................................................................................................... 150

VOIP TRUNKS .................................................................................................................................. 153

DIRECT OUTWARD DIALING (DOD) ....................................................................................... 164

SLA STATION .......................................................................................... 166

CREATE/EDIT SLA STATION ........................................................................................................... 166

SAMPLE CONFIGURATION ............................................................................................................ 167

CALL ROUTES ........................................................................................ 170

OUTBOUND ROUTES ..................................................................................................................... 170


INBOUND ROUTES ......................................................................................................................... 172

INBOUND RULE CONFIGURATIONS ....................................................................................... 173

INBOUND ROUTE: PREPEND EXAMPLE ................................................................................ 174

BLACKLIST CONFIGURATIONS .............................................................................................. 175

CONFERENCE BRIDGE ......................................................................... 178

CONFERENCE BRIDGE CONFIGURATIONS .......................................................................... 178

JOIN A CONFERENCE CALL .................................................................................................... 180

INVITE OTHER PARTIES TO JOIN CONFERENCE ................................................................. 180

DURING THE CONFERENCE ................................................................................................... 181

RECORD CONFERENCE ......................................................................................................... 182

IVR............................................................................................................ 184

CONFIGURE IVR ............................................................................................................................. 184

CREATE CUSTOM PROMPT ........................................................................................................... 186

RECORD NEW CUSTOM PROMPT ......................................................................................... 186

UPLOAD CUSTOM PROMPT ................................................................................................... 187

LANGUAGE SETTINGS FOR VOICE PROMPT ..................................... 188

DOWNLOAD AND INSTALL VOICE PROMPT PACKAGE .............................................................. 188

VOICEMAIL .............................................................................................. 190

CONFIGURE VOICEMAIL ................................................................................................................ 190

ACCESS VOICEMAIL ....................................................................................................................... 191

VOICEMAIL EMAIL SETTINGS ........................................................................................................ 192

CONFIGURE VOICEMAIL GROUP .................................................................................................. 193

RING GROUP .......................................................................................... 196

CONFIGURE RING GROUP ............................................................................................................ 196

REMOTE EXTENSION IN RING GROUP ........................................................................................ 198

PAGING AND INTERCOM GROUP ......................................................... 202

CONFIGURE PAGING/INTERCOM GROUP ................................................................................... 202

CALL QUEUE .......................................................................................... 204

CONFIGURE CALL QUEUE ............................................................................................................. 204

EXTENSION GROUPS ............................................................................ 208

CONFIGURE EXTENSION GROUPS .............................................................................................. 208

USING EXTENSION GROUPS ........................................................................................................ 209

PICKUP GROUPS ................................................................................... 210


CONFIGURE PICKUP GROUPS ..................................................................................................... 210

CONFIGURE PICKUP FEATURE CODE ......................................................................................... 210

MUSIC ON HOLD .................................................................................... 212

FAX/T.38 ................................................................................................... 214

CONFIGURE FAX/T.38 ..................................................................................................................... 214

SAMPLE CONFIGURATION TO RECEIVE FAX FROM PSTN LINE ............................................... 215

SAMPLE CONFIGURATION FOR FAX-TO-EMAIL .......................................................................... 217

ASTERISK MANAGER INTERFACE (RESTRICTED ACCESS) ............ 218

BUSY CAMP-ON ..................................................................................... 220

FOLLOW ME ............................................................................................ 222

ONE-KEY DIAL ........................................................................................ 224

DISA ......................................................................................................... 226

CALLBACK FEATURE ............................................................................ 228

BLF AND EVENT LIST ............................................................................ 230

BLF ................................................................................................................................................... 230

EVENT LIST ..................................................................................................................................... 230

DIAL BY NAME ........................................................................................ 234

DIAL BY NAME CONFIGURATION .................................................................................................. 234

ACTIVE CALLS AND MONITOR ............................................................. 238

ACTIVE CALLS STATUS .................................................................................................................. 238

HANG UP ACTIVE CALLS ............................................................................................................... 239

CALL MONITOR ............................................................................................................................... 239

CALL FEATURES .................................................................................... 242

FEATURE CODES ............................................................................................................................ 242

CALL RECORDING .......................................................................................................................... 246

CALL PARK ....................................................................................................................................... 247

PARK A CALL ............................................................................................................................ 247

RETRIEVE THE PARKED CALL ............................................................................................... 247

ENABLE SPY .................................................................................................................................... 247


INTERNAL OPTIONS .............................................................................. 250

INTERNAL OPTIONS/GENERAL ..................................................................................................... 250

INTERNAL OPTIONS/JITTER BUFFER .......................................................................................... 252

INTERNAL OPTIONS/RTP SETTINGS ............................................................................................ 252

INTERNAL OPTIONS/PORTS CONFIG ........................................................................................... 253

INTERNAL OPTIONS/STUN MONITOR .......................................................................................... 254

INTERNAL OPTIONS/PAYLOAD ...................................................................................................... 255

IAX SETTINGS ......................................................................................... 256

IAX SETTINGS/GENERAL ............................................................................................................... 256

IAX SETTINGS/REGISTRATION ..................................................................................................... 256

IAX SETTINGS/STATIC DEFENSE .................................................................................................. 257

SIP SETTINGS ......................................................................................... 258

SIP SETTINGS/GENERAL ............................................................................................................... 258

SIP SETTINGS/MISC ....................................................................................................................... 259

SIP SETTINGS/SESSION TIMER .................................................................................................... 259

SIP SETTINGS/TCP AND TLS ......................................................................................................... 260

SIP SETTINGS/NAT ......................................................................................................................... 261

SIP SETTINGS/TOS ......................................................................................................................... 261

VALUE-ADDED FEATURES ................................................................... 264

FAX SENDING .................................................................................................................................. 264

ANNOUNCEMENTS CENTER ......................................................................................................... 265

ANNOUNCEMENTS CENTER SETTINGS ............................................................................... 265

GROUP SETTINGS ................................................................................................................... 266

STATUS AND REPORTING ..................................................................... 268

PBX STATUS .................................................................................................................................... 268

TRUNKS .................................................................................................................................... 268

EXTENSIONS ............................................................................................................................ 269

QUEUES .................................................................................................................................... 270

CONFERENCE ROOMS ........................................................................................................... 271

INTERFACES STATUS .............................................................................................................. 272

PARKING LOT ........................................................................................................................... 273

SYSTEM STATUS ............................................................................................................................ 274

GENERAL .................................................................................................................................. 274

NETWORK ................................................................................................................................. 274

STORAGE USAGE .................................................................................................................... 275

RESOURCE USAGE ................................................................................................................. 275


SYSTEM EVENTS ............................................................................................................................ 276

ALERT EVENTS LIST ................................................................................................................ 276

ALERT LOG ............................................................................................................................... 278

ALERT CONTACT ..................................................................................................................... 280

CDR .................................................................................................................................................. 280

DOWNLOADED CDR FILE ........................................................................................................ 283

STATISTICS ............................................................................................................................... 284

RECORDING FILES .................................................................................................................. 285

API CONFIGURATION .............................................................................................................. 286

UPGRADING AND MAINTENANCE ....................................................... 288

UPGRADING .................................................................................................................................... 288

UPGRADING VIA NETWORK ................................................................................................... 288

UPGRADING VIA LOCAL UPLOAD .......................................................................................... 289

NO LOCAL FIRMWARE SERVERS .......................................................................................... 291

BACKUP ........................................................................................................................................... 291

BACKUP/RESTORE .................................................................................................................. 291

DATA SYNC ............................................................................................................................... 293

RESTORE CONFIGURATION FROM BACKUP FILE ............................................................... 294

CLEANER ......................................................................................................................................... 295

RESET AND REBOOT ..................................................................................................................... 297

SYSLOG ........................................................................................................................................... 297

TROUBLESHOOTING ...................................................................................................................... 298

ETHERNET CAPTURE .............................................................................................................. 298

IP PING ...................................................................................................................................... 299

TRACEROUTE .......................................................................................................................... 299

ANALOG RECORD TRACE ...................................................................................................... 300

SERVICE CHECK ...................................................................................................................... 301

REMOTE ACCESS ........................................................................................................................... 301

SSH ACCESS ............................................................................................................................ 301

EXPERIENCING THE UCM6100 SERIES IP PBX .................................. 304


Table of Tables


Table 1: Technical Specifications ................................................................................................................ 28

Table 2: UCM6102/UCM6104 Equipment Packaging ................................................................................ 31

Table 3: LCD Menu Options ....................................................................................................................... 38

Table 4: UCM6102/UCM6104 LED INDICATORS ...................................................................................... 39

Table 5: UCM6108/UCM6116 LED INDICATORS ...................................................................................... 40

Table 6: User Management - Create New User ......................................................................................... 47

Table 7: Operation Log Column Header ..................................................................................................... 50

Table 8: Change Binding Email option ....................................................................................................... 51

Table 9: UCM6102 Network Settings->Basic Settings ............................................................................... 52

Table 10: UCM6104 Network Settings->Basic Settings ............................................................................. 54

Table 11: UCM6108/UCM6116 Network Settings->Basic Settings ............................................................. 55

Table 12: UCM6100 Network Settings->802.1X ......................................................................................... 60

Table 13: UCM6100 Network Settings->Static Routes ............................................................................... 61

Table 14: UCM6102 Network Settings->Port Forwarding ........................................................................... 63

Table 15: UCM6100 Firewall->Static Defense->Current Service ............................................................... 67

Table 16: Typical Firewall Settings ............................................................................................................. 67

Table 17: Firewall Rule Settings ................................................................................................................. 68

Table 18: UCM6100 Firewall Dynamic Defense ......................................................................................... 69

Table 19: Fail2Ban Settings ........................................................................................................................ 70

Table 20: HTTP Server Settings ................................................................................................................. 79

Table 21: Email Settings ............................................................................................................................. 79

Table 22: Time Auto Updating .................................................................................................................... 81

Table 23: Create New Office Time .............................................................................................................. 83

Table 24: Create New Holiday .................................................................................................................... 85

Table 25: Auto Provision Settings ............................................................................................................... 94

Table 26: Global Policy Parameters: Localization ...................................................................................... 97

Table 27: Global Policy Parameters: Phone Settings ................................................................................. 97

Table 28: Global Policy Parameters: Contact List ...................................................................................... 98

Table 29: Global Policy Parameters: Maintenance ................................................................................... 100

Table 30: Global Policy Parameters: Network Settings ............................................................................ 102

Table 31: Global Policy Parameters: Customization ................................................................................ 102

Table 32: Create New Template ............................................................................................................... 104

Table 33: Create New Model Template ..................................................................................................... 106

Table 34: SIP Extension Configuration Parameters - Basic Settings ....................................................... 121

Table 35: SIP Extension Configuration Parameters - Media .................................................................... 123

Table 36: SIP Extension Configuration Parameters - Features ................................................................ 124

Table 37: SIP Extension Configuration Parameters - Specific Time ......................................................... 127

Table 38: IAX Extension Configuration Parameters - Basic Settings ....................................................... 127


Table 39: IAX Extension Configuration Parameters - Media .................................................................... 128

Table 40: IAX Extension Configuration Parameters - Features ................................................................ 129

Table 41: IAX Extension Configuration Parameters - Specific Time ......................................................... 131

Table 42: FXS Extension Configuration Parameters - Basic Settings ...................................................... 132

Table 43: FXS Extension Configuration Parameters - Media ................................................................... 133

Table 44: FXS Extension Configuration Parameters - Features ............................................................... 134

Table 45: FXS Extension Configuration Parameters - Specific Time ....................................................... 136

Table 46: Batch Add SIP Extension Parameters ...................................................................................... 136

Table 47: Batch Add IAX Extension Parameters ...................................................................................... 139

Table 48: Analog Trunk Configuration Parameters ................................................................................... 147

Table 49: PSTN Detection for Analog Trunk ............................................................................................. 153

Table 50: Create New SIP Trunk .............................................................................................................. 154

Table 51: SIP Register Trunk Configuration Parameters .......................................................................... 155

Table 52: SIP Peer Trunk Configuration Parameters ................................................................................ 158

Table 53: Create New IAX Trunk .............................................................................................................. 161

Table 54: IAX Register Trunk Configuration Parameters .......................................................................... 161

Table 55: IAX Peer Trunk Configuration Parameters ............................................................................... 162

Table 56: SLA Station Configuration Parameters ..................................................................................... 166

Table 57: Outbound Route Configuration Parameters ............................................................................. 170

Table 58: Inbound Rule Configuration Parameters .................................................................................. 173

Table 59: Conference Bridge Configuration Parameters .......................................................................... 178

Table 60: Conference Caller IVR Menu .................................................................................................... 181

Table 61: IVR Configuration Parameters .................................................................................................. 184

Table 62: Voicemail Settings .................................................................................................................... 190

Table 63: Voicemail IVR Menu ................................................................................................................. 191

Table 64: Voicemail Email Settings .......................................................................................................... 192

Table 65: Voicemail Group Settings ......................................................................................................... 194

Table 66: Ring Group Parameters ............................................................................................................ 196

Table 67: Paging/Intercom Group Configuration Parameters ................................................................... 202

Table 68: Call Queue Configuration Parameters ...................................................................................... 204

Table 69: FAX/T.38 Settings ..................................................................................................................... 214

Table 70: Follow Me Settings ................................................................................................................... 223

Table 71: Follow Me Options .................................................................................................................... 223

Table 72: DISA Settings ............................................................................................................................ 226

Table 73: Callback Configuration Parameters .......................................................................................... 228

Table 74: Event List Settings .................................................................................................................... 230

Table 75: UCM6100 Feature Codes ......................................................................................................... 242

Table 76: Internal Options/General ........................................................................................................... 250

Table 77: Internal Options/Jitter Buffer ..................................................................................................... 252

Table 78: Internal Options/RTP Settings .................................................................................................. 252

Table 79: Internal Options/Ports Config .................................................................................................... 253


Table 80: Internal Options/STUN Monitor ................................................................................................. 254

Table 81: Internal Options/Payload .......................................................................................................... 255

Table 82: IAX Settings/General ................................................................................................................ 256

Table 83: IAX Settings/Registration .......................................................................................................... 256

Table 84: IAX Settings/Static Defense ...................................................................................................... 257

Table 85: SIP Settings/General ................................................................................................................ 258

Table 86: SIP Settings/Misc ...................................................................................................................... 259

Table 87: SIP Settings/Session Timer ...................................................................................................... 259

Table 88: SIP Settings/TCP and TLS ....................................................................................................... 260

Table 89: SIP Settings/NAT ...................................................................................................................... 261

Table 90: SIP Settings/ToS ....................................................................................................................... 261

Table 91: Announcements Center Settings .............................................................................................. 265

Table 92: Group Settings .......................................................................................................................... 266

Table 93: Trunk Status .............................................................................................................................. 268

Table 94: Extension Status ....................................................................................................................... 270

Table 95: Agent Status ............................................................................................................................. 271

Table 96: Interface Status Indicators ........................................................................................................ 272

Table 97: Parking Lot Status .................................................................................................................... 273

Table 98: System Status->General ........................................................................................................... 274

Table 99: System Status->Network .......................................................................................................... 274

Table 100: CDR Filter Criteria .................................................................................................................. 280

Table 101: CDR Statistics Filter Criteria ................................................................................................... 285

Table 102: API Configuration Files ........................................................................................................... 286

Table 103: Network Upgrade Configuration ............................................................................................. 288

Table 104: Data Sync Configuration ......................................................................................................... 293

Table 105: Cleaner Configuration ............................................................................................................. 296


Table of Figures


Figure 1: UCM6102/UCM6104/UCM6108/UCM6116 ................................................................................. 27

Figure 2: UCM6102 Front View .................................................................................................................. 31

Figure 3: UCM6102 Back View .................................................................................................................. 32






Figure 9: UCM6116 Back View ................................................................................................................... 34

Figure 10: UCM6116 Web GUI Login Page ............................................................................................... 40

Figure 11: UCM6100 Web GUI Language .................................................................................................. 42

Figure 12: User Management Page Display .............................................................................................. 45

Figure 13: Create New User ....................................................................................................................... 47

Figure 14: User Management – New Users ............................................................................................... 47

Figure 15: Edit User Information by Super Admin ...................................................................................... 48

Figure 16: User Portal Login ...................................................................................................................... 48

Figure 17: User Portal Layout .................................................................................................................... 49

Figure 18: Multiple User Operation Error Prompt ....................................................................................... 49

Figure 19: Operation Logs .......................................................................................................................... 50

Figure 20: Operation Logs Filter ................................................................................................................. 51

Figure 21: Change Binding Email ............................................................................................................... 51

Figure 22: UCM6102 Network Interface Method: Route ............................................................................ 57

Figure 23: UCM6102 Network Interface Method: Switch ........................................................................... 58

Figure 24: UCM6102 Network Interface Method: Dual .............................................................................. 59

Figure 25: UCM6100 Using 802.1X as Client ............................................................................................ 59

Figure 26: UCM6100 Using 802.1X EAP-MD5 .......................................................................................... 60

Figure 27: UCM6104 Static Route Sample ................................................................................................ 62

Figure 28: UCM6104 Static Route Configuration ....................................................................................... 62

Figure 29: UCM6102 Port Forwarding Configuration ................................................................................. 64

Figure 30: GXP2160 Web Access Using UCM6102 Port Forwarding ........................................................ 64

Figure 31: Register Domain Name on noip.com ........................................................................................ 65

Figure 32: UCM6100 DDNS Setting ........................................................................................................... 66

Figure 33: Using Domain Name to Connect to UCM6100 ......................................................................... 66

Figure 34: Create New Firewall Rule ......................................................................................................... 68

Figure 35: Configure Dynamic Defense ..................................................................................................... 70

Figure 36: LDAP Server Configurations ..................................................................................................... 72

Figure 37: Default LDAP Phonebook DN ................................................................................................... 73

Figure 38: Default LDAP Phonebook Attributes ......................................................................................... 73


Figure 39: LDAP Server: LDAP Phonebook ............................................................................................... 74

Figure 40: Add LDAP Phonebook ............................................................................................................... 74

Figure 41: Edit LDAP Phonebook ............................................................................................................... 75

Figure 42: Import Phonebook ..................................................................................................................... 75

Figure 43: Phonebook CSV File Format .................................................................................................... 76

Figure 44: LDAP Phonebook After Import .................................................................................................. 76

Figure 45: Export Selected LDAP Phonebook ........................................................................................... 77

Figure 46: LDAP Client Configurations ....................................................................................................... 77

Figure 47: GXP2200 LDAP Phonebook Configuration ............................................................................... 78

Figure 48: UCM6100 Email Settings .......................................................................................................... 80

Figure 49: Set Time Manually ..................................................................................................................... 82

Figure 50: Create New Office Time ............................................................................................................ 83

Figure 51: Settings->Time Settings->Office Time ....................................................................................... 84

Figure 52: Create New Holiday .................................................................................................................. 85

Figure 53: Settings->Time Settings->Holiday ............................................................................................. 86

Figure 54: Settings->Recordings Storage .................................................................................................. 87

Figure 55: Recordings Storage Prompt Information ................................................................................... 88

Figure 56: Recording Storage Category ..................................................................................................... 88

Figure 57: Login Timeout Settings .............................................................................................................. 89

Figure 58: Zero Config Configuration Architecture for End Point Device ................................................... 92

Figure 59: UCM6100 Zero Config .............................................................................................................. 93

Figure 60: Auto Provision Settings ............................................................................................................. 94

Figure 61: Auto Discover ............................................................................................................................ 95

Figure 62: Discovered Devices .................................................................................................................. 96

Figure 63: Global Policy Categories ........................................................................................................... 97

Figure 64: Edit Global Template ............................................................................................................... 105

Figure 65: Edit Model Template ................................................................................................................ 107

Figure 66: Template Management ............................................................................................................ 108

Figure 67: Upload Model Template Manually ........................................................................................... 109

Figure 68: Create New Device ................................................................................................................. 110

Figure 69: Manage Devices ..................................................................................................................... 110

Figure 70: Edit Device .............................................................................................................................. 111

Figure 71: Edit Customize Device Settings .............................................................................................. 112

Figure 72: Add P Value in Customize Device Settings ............................................................................. 113

Figure 73: Modify Selected Devices - Same Model ................................................................................. 114

Figure 74: Modify Selected Devices - Different Models ............................................................................ 115

Figure 75: Device List in Zero Config ....................................................................................................... 116

Figure 76: Zero Config Sample - Global Policy ........................................................................................ 117

Figure 77: Zero Config Sample - Device Preview 1 ................................................................................. 118




Figure 80: Create New Device ................................................................................................................. 121

Figure 81: Manage Extensions ................................................................................................................. 142

Figure 82: Export Extensions ................................................................................................................... 143

Figure 83: Import Extensions ................................................................................................................... 143

Figure 84: Email To User - Prompt Information ........................................................................................ 144

Figure 85: Account Registration Information and QR Code ..................................................................... 145

Figure 86: LDAP Client Information and QR Code ................................................................................... 145

Figure 87: UCM6100 FXO Tone Settings ................................................................................................. 151

Figure 88: UCM6100 PSTN Detection ..................................................................................................... 151

Figure 89: UCM6100 PSTN Detection: Auto Detect ................................................................................. 152

Figure 90: UCM6100 PSTN Detection: Semi-Auto Detect ....................................................................... 152

Figure 91: DOD extension selection ......................................................................................................... 165

Figure 92: Edit DOD ................................................................................................................................. 165

Figure 93: SLA Station ............................................................................................................................. 166

Figure 94: Enable SLA Mode for Analog Trunk ........................................................................................ 168

Figure 95: Analog Trunk with SLA Mode Enabled .................................................................................... 168

Figure 96: SLA Example - SLA Station ..................................................................................................... 168

Figure 97: SLA Example - MPK Configuration ......................................................................................... 169

Figure 98: Inbound Route feature: Prepend ............................................................................................. 175

Figure 99: Blacklist Configuration Parameters ......................................................................................... 176

Figure 100: Blacklist csv File .................................................................................................................... 176

Figure 101: Conference Invitation From Web GUI ................................................................................... 180

Figure 102: Conference Recording .......................................................................................................... 183

Figure 103: Click on Prompt to Create IVR Prompt ................................................................................. 186

Figure 104: Record New Custom Prompt ................................................................................................ 186

Figure 105: Upload Custom Prompt ......................................................................................................... 187

Figure 106: Language Settings for Voice Prompt ..................................................................................... 188

Figure 107: Voice Prompt Package List ................................................................................................... 189

Figure 108: New Voice Prompt Language Added .................................................................................... 189

Figure 109: Voicemail Email Settings ....................................................................................................... 193

Figure 110: Voicemail Group .................................................................................................................... 194

Figure 111: Ring Group ............................................................................................................................ 196

Figure 112: Ring Group Configuration ...................................................................................................... 198

Figure 113: Sync LDAP Server option ...................................................................................................... 199

Figure 114: Manually Sync LDAP Server ................................................................................................. 199

Figure 115: Ring Group Remote Extension .............................................................................................. 200

Figure 116: Paging/Intercom Group ......................................................................................................... 202

Figure 117: Page/Intercom Group Settings .............................................................................................. 203

Figure 118: Call Queue ............................................................................................................................. 204

Figure 119: Agent Login Settings .............................................................................................................. 207

Figure 120: Edit Extension Group ............................................................................................................ 208


Figure 121: Select Extension Group in Outbound Route ......................................................................... 209

Figure 122: Edit Pickup Group ................................................................................................................. 210

Figure 123: Edit Pickup Feature Code ..................................................................................................... 211

Figure 124: Music On Hold Default Class ................................................................................................ 212

Figure 125: Configure Analog Trunk without Fax Detection ..................................................................... 215

Figure 126: Configure Extension for Fax Machine ................................................................................... 216

Figure 127: Configure Inbound Rule for Fax ............................................................................................ 216

Figure 128: Create Fax Extension ............................................................................................................ 217

Figure 129: Inbound Route to Fax Extension ........................................................................................... 217

Figure 130: Create Follow Me .................................................................................................................. 222

Figure 131: Edit Follow Me ....................................................................................................................... 222

Figure 132: Configure One-Key Dial ........................................................................................................ 224

Figure 133: One-Key Dial Destinations .................................................................................................... 225

Figure 134: Create New DISA .................................................................................................................. 226

Figure 135: Create New Event List .......................................................................................................... 231

Figure 136: Create Dial By Name Group ................................................................................................. 234

Figure 137: Dial By Name Group In IVR Key Pressing Events ................................................................ 235

Figure 138: Dial By Name Group In Inbound Rule ................................................................................... 236

Figure 139: Configure Extension First Name and Last Name .................................................................. 237

Figure 140: Status->PBX Status->Active Calls - Ringing ......................................................................... 238

Figure 141: Status->PBX Status->Active Calls – Call Established .......................................................... 238

Figure 142: Configure to Monitor an Active Call ....................................................................................... 239

Figure 143: Download Recording File from CDR Page ............................................................................ 246

Figure 144: Download Recording File from Recording Files Page .......................................................... 247

Figure 145: FXS Ports Signaling Preference ........................................................................................... 253

Figure 146: FXO Ports ACIM Settings ...................................................................................................... 253

Figure 147: Fax Sending in Web UI ......................................................................................................... 264

Figure 148: Announcements Center ......................................................................................................... 265

Figure 149: Announcements Center Group Configuration ....................................................................... 266

Figure 150: Announcements Center Code Configuration ......................................................................... 267

Figure 151: Announcements Center Example .......................................................................................... 267

Figure 152: Status->PBX Status .............................................................................................................. 268

Figure 153: Trunk Status .......................................................................................................................... 268

Figure 154: Extension Status ................................................................................................................... 269

Figure 155: Queue Status ........................................................................................................................ 271

Figure 156: Conference Room Status ...................................................................................................... 271

Figure 157: UCM6116 Interfaces Status .................................................................................................. 272

Figure 158: Parking Lot Status ................................................................................................................. 273

Figure 159: System Status->Storage Usage ............................................................................................ 275

Figure 160: System Status->Resource Usage ......................................................................................... 276

Figure 161: System Events->Alert Events Lists: Disk Usage ................................................................... 277


Figure 162: System Events->Alert Events Lists: Memory Usage ............................................................. 277

Figure 163: System Events->Alert Events Lists: System Reboot ............................................................. 278

Figure 164: System Events->Alert Events Lists: System Crash ............................................................... 278

Figure 165: System Events->Alert Log ..................................................................................................... 279

Figure 166: Filter for Alert Log .................................................................................................................. 279

Figure 167: CDR Filter ............................................................................................................................. 280

Figure 168: Call Report ............................................................................................................................ 281

Figure 169: Call Report Entry with Audio Recording File ......................................................................... 282

Figure 170: Automatic Download Settings ............................................................................................... 282

Figure 171: Downloaded CDR File Sample - Call To Shows "s" .............................................................. 283

Figure 172: Downloaded CDR File Sample - Source Channel and Dest Channel 1 ................................ 283



Figure 175: CDR Statistics ....................................................................................................................... 285

Figure 176: CDR->Recording Files .......................................................................................................... 286

Figure 177: Network Upgrade .................................................................................................................. 288

Figure 178: Local Upgrade ....................................................................................................................... 289

Figure 179: Upgrading Firmware Files ..................................................................................................... 290

Figure 180: Reboot UCM6100 ................................................................................................................. 290

Figure 181: Create New Backup .............................................................................................................. 292

Figure 182: Backup / Restore ................................................................................................................... 292

Figure 183: Local Backup ......................................................................................................................... 293

Figure 184: Data Sync .............................................................................................................................. 293

Figure 185: Restore UCM6100 from Backup File .................................................................................... 295

Figure 186: Cleaner .................................................................................................................................. 296

Figure 187: Reset and Reboot ................................................................................................................. 297

Figure 188: Ethernet Capture ................................................................................................................... 298

Figure 189: PING ..................................................................................................................................... 299

Figure 190: Traceroute ............................................................................................................................. 300

Figure 191: Troubleshooting Analog Trunks ............................................................................................. 301

Figure 192: Service Check ....................................................................................................................... 301

Figure 193: SSH Access .......................................................................................................................... 302


GNU GPL INFORMATION

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

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

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

Grandstream GNU GPL related source code can be downloaded from Grandstream web site from: http://www.grandstream.com/support/faq/gnu_gpl


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 1.0.9.26

•

Added Announcements Center [ANNOUNCEMENTS CENTER]


•

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]


•

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 58: 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. [USER PORTAL]

•

Added support to send Fax via web UI. [FAX SENDING]

•

Added "Enable LDAP" option to skip the extension from UCM default LDAP phonebook. [Table 34:

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 100: 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 36: SIP Extension

Configuration Parameters - Features]


•

Added option to send P-Asserted-Identity header in SIP Register Trunk. [Table 51: 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 51: SIP Register Trunk

Configuration Parameters]

•

Added LDAP Client support [LDAP CLIENT CONFIGURATIONS]

•

Added option to specify the chronological order of voice mails. [Table 62: Voicemail Settings]

•

Added option to configure whether to skip pressing 1/2 to accept or reject calls from Follow Me.

[Table 70: Follow Me Settings]

•

Added option to specify port range in Port Forwarding configuration. [Table 14: 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 29: 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. [Table 52: SIP Peer Trunk Configuration Parameters]

•

Improved Automatic Download CDR result format. [CDR]


•

Added recording files store management. [RECORDINGS STORAGE]

•

Added AMI access (restricted). [ASTERISK MANAGER INTERFACE (RESTRICTED ACCESS)]

•

Added LDAP Phonebook Import/Export support. [LDAP PHONEBOOK]

•

Added Callback feature. [CALLBACK FEATURE]

•

Added Busy Camp-on (Call Completion) feature. [BUSY CAMP-ON]

•

Added external number calling via extension. [Table 36: SIP Extension Configuration Parameters -

Features]

•

Added end point firmware management support to allow end point device to upgrade via

UCM6100 local storage or USB media. [Table 29: Global Policy Parameters: Maintenance]

•

Added link to access phone web GUI in Zero Config. [MANAGE DEVICES]

•

Added model update support to download model templates for surveillance and GSwave. [MODEL

UPDATE]

•

Added filter for searching in Alert Log. [ALERT LOG]

•

Added DAHDI Out Line Selection. [Table 48: Analog Trunk Configuration Parameters]

•

Added automatic CDR download. [CDR]

•

Added multi-user management and concurrent user login support. [USER MANAGEMENT]

•

Added operation log. [OPERATION LOG]


•

Added option to specify web access session timer. [LOGIN TIMEOUT SETTINGS]

•

Added call monitor/barge-in support. [ACTIVE CALLS AND MONITOR]

•

Added GSwave QR code support. [EMAIL TO USER]

•

Added ability to add customized Pvalue on Model Template. [MODEL TEMPLATES]

•

Updated maximum length of web UI login password to 30 characters. [CHANGE PASSWORD]

•

Added Fax gateway support for Fax mode on FXS and FXO port. [Table 43: FXS Extension

Configuration Parameters - Media] [Table 48: Analog Trunk Configuration Parameters]

•

Added “NAT” option and “Send PPI Header” option in SIP register trunk configuration. [Table 51:

SIP Register Trunk Configuration Parameters]

•

Added REC API (access for call recording files). [API CONFIGURATION]

•

Added “Synchronize All Data” option in Data Sync. [DATA SYNC]

•

Added “Delect Selected Sound Files” option for Music On Hold files. [MUSIC ON HOLD]


•

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

•

Added new architecture for auto provisioning via Zero Config. [


PROVISIONING]

•

Added Alert-info configuration for distinctive ringing on inbound route. [Table 58: Inbound Rule


•

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

•

Added Follow Me support. [FOLLOW ME]

•

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

•

Added FXO Dial Delay option for analog trunk. [Table 48: Analog Trunk Configuration Parameters]

•

Added Music On Hold selection per extension. [EXTENSIONS]

•

Added Call Forward Time Condition per extension. [EXTENSIONS]

•

Added custom prompt support for ring group announcement. [Table 66: Ring Group Parameters]

•

Added custom prompt support for paging/intercom group announcement. [Table 67:

Paging/Intercom Group Configuration Parameters]

•

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

•

Added DTMF configuration per trunk. [VOIP TRUNKS]

•

Added call queue Music On Hold customization. [Table 68: Call Queue Configuration Parameters]

•

Added call queue destination support. [Table 68: Call Queue Configuration Parameters]

•

Added batch add blacklist file support. [BLACKLIST CONFIGURATIONS]


•

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


•

Added static routes function. [STATIC ROUTES]

•

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

•

Added option to disable extension/trunk.

•

Added TEL URL support for extension/trunk.

•

Added option to dial trunk password per extensions.

•

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

EXTENSIONS]

•

Added option "Need Registration" for SIP register trunk.

•

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

•

Added Dial By Name. [DIAL BY NAME]

•

Added voicemail password and Email address for voicemail group extension.

•

Added auto record support for ring group and call queue.

•

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

•

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


•

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

•

Added activity calls status on web UI status page.

•

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]


•

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

•

Added "Direct Dial Voicemail Prefix" feature code back. [Table 75: UCM6100 Feature Codes]

•

Changed valid range for option "Current Disconnect Threshold". [Table 48: Analog Trunk



•

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

RESET and RE-CONFIGURED MANUALLY.

•

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

•

Updated LDAP configuration example. [LDAP SERVER]

•

Added "Enable Filter Source Caller ID" and "Custom Dynamic Route" options for outbound route settings. [Table 57: Outbound Route Configuration Parameters]

•

Added more language support for voice prompt. [LANGUAGE SETTINGS FOR VOICE PROMPT]

•

Added "Ring Group Destination" for ring group configuration. [Table 66: Ring Group Parameters]

•

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

•

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

•

Added BLF function description. [BLF AND EVENT LIST]

•

Updated default extension range. [Table 76: Internal Options/General]

•

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


•

Asterisk updated to version 1.8.23.1.

•

Added DID routing support for incoming calls. [Table 50: Create New SIP Trunk]

•

Added DOD routing support. [DIRECT OUTWARD DIALING (DOD)]

•

Added GXP one-button Voicemail access. [Table 34: SIP Extension Configuration Parameters]


•

Added option "Skip voicemail password verification" on extension edit page. [Table 34: SIP

Extension Configuration Parameters]

•

Added Hot-Desking Support. [Table 34: SIP Extension Configuration Parameters]

•

Added one-button on-demand call recording for GXP

•

Add new option to enable or disable "FXS TISS Override" on Hardware Config page. [Table 79:

Internal Options/Ports Config]

•

Added more modes for FXS Two-Wire Impedance Synthesis

•

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

•

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

•

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

•

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

•

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

[AUTO PROVISIONING]

•

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

[Figure 164: System Events->Alert Events Lists: System Crash]

•

Add "Keep Trunk CID" option for VoIP trunks, and keep the priorities: DOD->Extension

CallerID->Trunk CallerID->Global CallerID. [Table 50: Create New SIP Trunk]


•

Added Fail2Ban support for SIP authentication. [FAIL2BAN]

•

Added voice prompt "Language" selection and "Auto Record" option for extension. [Table 34: SIP

Extension Configuration Parameters] [Table 46: Batch Add SIP Extension Parameters]

•

Added "Auto Record" option for trunk.

•

Added "Digit Timeout" option and voice prompt "Language" selection for IVR. [Table 61: IVR


•

Added "Direct Dial Voicemail Prefix" feature code to directly dial or transfer to extension's voicemail. [Table 75: UCM6100 Feature Codes]

•

Added "Enforce Strong Passwords" option. [Table 76: Internal Options/General]

•

Added FXS MWI Mode. [Table 79: Internal Options/Ports Config]

•

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

•

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


•

Added weight information. [Table 1: Technical Specifications]

•

Added NTP server support. [NTP SERVER]

•

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

•

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


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

•

Updated LDAP client configurations information.

•

Added sample Email settings. [EMAIL SETTINGS]

•

Added manual time settings. [TIME SETTINGS]

•

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

•

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

(NOTIFY)" in zero-config feature.

•

Added pickup group feature. [Table 75: UCM6100 Feature Codes]

•

Added PSTN detection instructions for "Auto Detect" and "Semi-auto Detect". [PSTN

DETECTION]

•

Added "Auth ID" option for SIP register trunk configuration. [Table 50: Create New SIP Trunk]

•

Added LDAP sync options for peer SIP trunk. [Table 50: Create New SIP Trunk]

•

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

•

Added DISA and Fax to IVR key press event options. [Table 61: IVR Configuration Parameters]

•

Added "Min Message Time" option in voicemail settings. [Table 62: Voicemail Settings]

•

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

•

Added DISA support for inbound route and IVR.

•

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

LIST]

•

Added feature code *0 for "Disconnect". [Table 75: UCM6100 Feature Codes]

•

Added feature code *8 for "Pickup Extension" in pickup group feature. [Table 75: UCM6100

Feature Codes]

•

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

76: Internal Options/General]

•

Added warning information for "Allow Guest Call" option to avoid potential security risk caused by misconfiguration. [Table 82: IAX Settings/General]

•

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


•

This is the initial version.



WELCOME

Thank you for purchasing Grandstream UCM6100 series IP PBX appliance. The UCM6100 series IP PBX is an innovative IP PBX appliance designed for small to medium business. Powered by an advanced hardware platform with robust system resources, the UCM6100 offers a highly versatile state-of-the-art

Unified Communication (UC) solution for converged voice, video, data, fax and video surveillance application needs. Incorporating industry-leading features and performance, the UCM6100 offers quick setup, deployment with ease and unrivaled reliability all at an unprecedented price point.

--------------------------------------------------------------------------------------------------------------------------------------------

Caution:

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

Warning:

Please do not use a different power adaptor with the UCM6100 as it may cause damage to the products and void the manufacturer warranty.

--------------------------------------------------------------------------------------------------------------------------------------------

This document is subject to change without notice. The latest electronic version of this user manual is available for download here:

http://www.grandstream.com/support

Reproduction or transmittal of the entire or any part, in any form or by any means, electronic or print, for any purpose without the express written permission of Grandstream Networks, Inc. is not permitted.



PRODUCT OVERVIEW

FEATURE HIGHTLIGHTS

•

1GHz ARM Cortex A8 application processor, large memory (512MB DDR RAM, 4GB NAND Flash), and dedicated high performance multi-core DSP array for advanced voice processing.

•

Integrated 2/4/8/16 PSTN trunk FXO ports, 2 analog telephone FXS ports with lifeline* capability in case of power outage, and up to 50 SIP trunk options.

•

Gigabit network port(s) with integrated PoE, USB, SD;; integrated NAT router with advanced QoS support (UCM6102 only).

•

Supports a wide range of popular voice codes (including G.711 A-law/U-law, G.722, G.723.1, G.726,

G.729A/B, iLBC (30ms only), GSM), video codec (including H.264, H.263, H.263+), and Fax (T.38).

•

Hardware DSP based 128ms-tail-length carrier-grade line echo cancellation (LEC).

•

Supports up to 500 SIP endpoint registration, up to 60 concurrent calls and up to 32 conference attendees.

•

Flexible dial plan, call routing, site peering, call recording.

•

Automated detection and provisioning of IP phones, video phones, ATA and other endpoints for easy deployment.

•

Hardware encryption accelerator to ensure strongest security protection using SRTP, TLS, and

HTTPS.

Firmware Version 1.0.9.26

Figure 1: UCM6102/UCM6104/UCM6108/UCM6116

UCM6100 Series IP PBX User Manual

Page 27 of 304

TECHNICAL SPECIFICATIONS

Voice-over-Packet

Capabilities

Voice and Fax Codecs

Video Codecs

QoS

Signaling and Control

DTMF Methods

Table 1: Technical Specifications

Interfaces

Analog Telephone FXS Ports 2 ports (both with lifeline capability in case of power outage)

PSTN Line FXO Ports

Network Interfaces

NAT Router

Peripheral Ports

LED Indicators

LCD Display

Reset Switch

•

UCM6102: 2 ports

•

UCM6104: 4 ports

•

UCM6108: 8 ports

•

UCM6116: 16 ports

•

UCM6102/6104: Dual 10M/100M/1000M RJ45 Ethernet ports with integrated PoE Plug (IEEE 802.3at-2009)

•

UCM6108/6116: Single 10M/100M/1000M RJ45 Ethernet port with integrated PoE Plug (IEEE 802.3at-2009)

Yes, UCM6102 only

USB, SD/SDHC (VFAT)

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

128x32 graphic LCD with DOWN and OK button

Yes

Voice/Video Capabilities

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;; T.38

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

Layer 3 QoS

Provisioning Protocol and

Plug-and-Play

Network Protocols

Disconnect Methods

In Audio, RFC2833, and SIP INFO

TFTP/HTTP/HTTPS, auto-discovery and auto-provisioning of

Grandstream IP endpoints via ZeroConfig (DHCP Option 66/multicast SIP

SUBSCRIBE/mDNS)

TCP/UDP/IP, RTP/RTCP, ICMP, ARP, DNS, DDNS, DHCP, NTP, TFTP,

SSH, HTTP/HTTPS, PPPoE, SIP (RFC3261), STUN, SRTP, TLS

Call Progress Tone, Polarity Reversal, Hook Flash Timing, Loop Current

Disconnect, Busy Tone


Security

Media SRTP, TLS, HTTPS, SSH

Physical

Universal Power Supply

Environmental

Dimensions

Weight

Mounting

•

Output: 12VDC, 1.5A

•

Input: 100-240VAC, 50-60Hz

•

Operating: 32 - 104 o

F / 0 - 40 o

C, 10-90% (non-condensing)

•

Storage: 14 - 140 o

F / -10 - 60 o

C

•

UCM6102/6104: 226mm (L) x 155mm (W) x 34.5mm (H)

•

UCM6108/6116: 440mm (L) x 185mm (W) x 44mm (H)

•

UCM6102: Unit weight 0.51kg, Package weight 0.94kg

•


•


•


•

UCM6102/6104: Wall mount and Desktop

•

UCM6108/6116: Rack mount and Desktop

Additional Features

Multi-language Support

Caller ID

Polarity Reversal/ Wink

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

GUI;; Customizable IVR to support any language

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

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

Call Center

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

Customizable Auto Attendant Up to 5 layers of IVR (Interactive Voice Response)

Concurrent Calls

Conference Bridges

Call Features

Compliance

•

UCM6102: Up to 30 simultaneous calls

•

UCM6104: Up to 45 simultaneous calls

•

UCM6108/6116: Up to 60 simultaneous calls

•

UCM6102/6104: Up to 3 password-protected conference bridges allowing up to 25 simultaneous PSTN or IP participants

•

UCM6108/6116: Up to 6 password-protected conference bridges allowing up to 32 simultaneous PSTN or IP participants

Call 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)

--------------------------------------------------------------------------------------------------------------------------------------------

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.

---------------------------------------------------------------------------------------------------------------------


INSTALLATION

Before deploying and configuring the UCM6100 series, the device needs to be properly powered up and

connected to network. This section describes detailed information on installation, connection and warranty policy of the UCM6100 series.

EQUIPMENT PACKAGING

Table 2: UCM6102/UCM6104 Equipment Packaging

Main Case

Power Adaptor

Ethernet Cable

Quick Installation Guide

GPL License

Yes (1)

Yes (1)

Yes (1)

Yes (1)

Yes (1)

CONNECT YOUR UCM6100

CONNECT THE UCM6102

Figure 2: UCM6102 Front View


Figure 3: UCM6102 Back View

To set up the UCM6102, follow the steps below:

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

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

3. Connect the 12V DC power adapter into the 12V DC power jack on the back of the UCM6102. Insert the main plug of the power adapter into a surge-protected power outlet.

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

5. Once the UCM6102 is successfully connected to network, the LED indicator for WAN in the front will

be in solid green and the LCD shows up the IP address.

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

Fax) to the FXS ports.



Firmware Version 1.0.9.26 UCM6100 Series IP PBX User Manual

Page 32 of 304



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




5. Once the UCM6104 is successfully connected to network, the LED indicator for LAN 1 in the front will

be in solid green and the LCD shows up the IP address.





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




5. Once the UCM6108 is successfully connected to network, the LED indicator for NETWORK in the front will be in solid green and the LCD shows up the IP address.










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





5. Once the UCM6116 is successfully connected to network, the LED indicator for NETWORK in the front will be in solid green and the LCD shows up the IP address.



SAFETY COMPLIANCES

The UCM6100 series IP PBX complies with FCC/CE and various safety standards. The UCM6100 power adapter is compliant with the UL standard. Use the universal power adapter provided with the UCM6100

package only. The manufacturer’s warranty does not cover damages to the device caused by unsupported power adapters.

WARRANTY

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

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

--------------------------------------------------------------------------------------------------------------------------------------------

Warning:

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

--------------------------------------------------------------------------------------------------------------------------------------------



GETTING STARTED

The UCM6100 series provides LCD interface, LED indication and web GUI configuration interface.

•

The LCD displays hardware, software and network information. Users could also navigate in the LCD menu for device information and basic network configuration.

•

The LED indication at the front of the device provides interface connection and activity status.

•

The web GUI gives users access to all the configurations and options for UCM6100 series setup.

This section provides step-by-step instructions on how to use the LCD menu, LED indicators and Web GUI

of the UCM6100 series. Once the basic settings are done, users could start making calls from UCM6100 extension registered on a SIP phone as described at the end of this section.

USE THE LCD MENU

•

Default LCD Display

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

•

Menu Navigation

Press the "Down" arrow key to browser different menu options. Press the "OK" button to select an entry.

Options].

•

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.


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

UCM6100 Series IP PBX User Manual Page 38 of 304

Web Info

SSH Switch

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

•

RTC Test Patterns

Select "2022-02-22 22:22" or "2011-01-11 11:11" to start the RTC

(Real-Time Clock) test pattern. Then check the system time from LCD idle screen by pressing "DOWN" button, or from web GUI->System

Status->General page. Reboot the device manually after the RTC test is done.

•

Hardware Testing

Select "Test SVIP" to perform SVIP test on the device. This is mainly for factory testing purpose which verifies the hardware connection inside the device. The diagnostic result will display in the LCD after the test is done.

•

Protocol: Web access protocol. HTTP or HTTPS. By default it's HTTPS

•

Port: Web access port number. By default it's 8089

•

Enable SSH: Enable SSH access.

•

Disable SSH: Disable SSH access.

By default the 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 Indicator

LAN

WAN

USB

SD

FXS (Phone/Fax)

FXO (Telco Line)

LED Status

Solid: Connected

Flashing: Data Transferring

OFF: Not Connected


LED

NETWORK

Table 5: UCM6108/UCM6116 LED INDICATORS

LED Status

Solid: Connected


ACT

USB

SD

Phone (FXS)

Line (FXO)

Solid: Connected

Flashing: Data Transferring


USE THE WEB GUI

ACCESS WEB GUI

The UCM6100 embedded Web server responds to HTTP/HTTPS GET/POST requests. Embedded HTML pages allow users to configure the device through a Web browser such as Microsoft IE, Mozilla Firefox,

Google Chrome and etc.


Figure 10: UCM6116 Web GUI Login Page


To access the Web GUI:

1. Connect the computer to the same network as the UCM6100.

2. Ensure the device is properly powered up and shows its IP address on the LCD.

3. Open a Web browser on the computer and enter the web GUI URL in the following format:

http(s)://IP-Address:Port where the IP-Address is the IP address displayed on the UCM6100 LCD.

By default, the protocol is HTTPS and the Port number is 8089.

For example, if the LCD shows 192.168.40.167, please enter the following in your web browser:

https://192.168.40.167:8089

4. Enter the administrator’s login and password to access the Web Configuration Menu. The default administrator's username and password is "admin" and "admin". It is highly recommended to change the default password after login for the first time.

--------------------------------------------------------------------------------------------------------------------------------------------

Note:

By default, the UCM6100 has "Redirect From Port 80" enabled. Therefore, if users type in the UCM6100 IP address in the web browser, the web page will be automatically redirected to the page using HTTPS and port 8089. For example, if the LCD shows 192.168.40.167, please enter 192.168.40.167 in your web

browser and the web page will be redirected to:

https://192.168.40.167:8089

The option "Redirect From Port 80" can be configured under the UCM6100 web GUI->Settings->HTTP

Server.

--------------------------------------------------------------------------------------------------------------------------------------------

WEB GUI CONFIGURATIONS

There are four main sections in the Web GUI for users to view the PBX status, configure and manage the

PBX.

•

Status: Displays PBX status, System Status, System Events and CDR.

•

PBX: To configure extensions, trunks, call routes, zero config for auto provisioning, call features, internal options, IAX settings and SIP settings.


•

Settings: To configure user management, network settings, firewall settings, change password,

LDAP Server, HTTP Server, Email Settings, Time Settings, NTP server, recording storage and login timeout.

•

Maintenance: To perform firmware upgrade, backup configurations, cleaner setup, reset/reboot, syslog setup and troubleshooting.

WEB GUI LANGUAGES

Currently the UCM6100 series web GUI supports 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

SAVE AND APPLY CHANGES

Click on "Save" button after configuring the web GUI options in one page. After saving all the changes,

make sure click on "Apply Changes" button on the upper right of the web page to submit all the changes. If the change requires reboot to take effect, a prompted message will pop up for you to reboot the device.

MAKE YOUR FIRST CALL


Power up the UCM6100 and your SIP end point phone. Connect both devices to the network. Then follow the steps below to make your first call.

1. Log in the UCM6100 web GUI, go to PBX->Basic/Call Routes->Extensions.

2. Click on "Create New SIP Extension" to create a new extension. You will need User ID, Password and

Voicemail Password information to register and use the extension later.

3. Register the extension on your phone with the SIP User ID, SIP server and SIP Password information.

The SIP server address is the UCM6100 IP address.

4. When your phone is registered with the extension, dial *97 to access the voicemail box. Enter the

Voicemail Password once you hear "Password" voice prompt.

5. Once successfully logged in to the voicemail, you will be prompted with the Voice Mail Main menu.

6. You are successfully connected to the PBX system now.



SYSTEM SETTINGS

This section explains configurations for system-wide parameters on the UCM6100. System settings are under “Settings” tag on UCM6100 web GUI. System settings include User Management, Network Settings,

Firewall, Change Password, LDAP server, HTTP server, Email settings, Time Settings, NTP Server,

Recordings Storage and Login Timeout settings.

USER MANAGEMENT

User management is on web GUI->Settings->User 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 UI using their extension number and password. All existing user accounts for web UI login will be

displayed on User Management page as shown in the following figure.

Figure 12: User Management Page Display

USER PRIVILEGES

Three privilege levels are supported:

•

Super Admin


- This is the highest privilege. Super Admin can access all pages on UCM6100 web GUI, change configuration for all options and execute all the operations.

- Super Admin can create, edit and delete one or more users with “Admin” privilege

- Super Admin can edit and delete one or more users with “Consumer” privilege

- Super Admin can view operation logs generated by all users.

- By default, the user account “admin” is configured with “Super Admin” privilege and it’s the only user with “Super Admin” privilege. The User Name and Privilege level cannot be changed or deleted.

- Super Admin could change its own login password on web UI->Settings->Change Password page.

- Super Admin could view operations done by all the users in web UI->Settings->User

Management->Operation Log.

•

Admin

- Users with “Admin” privilege can only be created by “Super Admin” user.

- “Admin” privilege users are not allowed to access the following pages:

Maintenance->Upgrade

Maintenance->Backup

Maintenance->Cleaner

Maintenance->Reset/Reboot

Settings->User Management->Operation Log

- “Admin” privilege users cannot create new users for login.

•

Consumer

- A user account for web UI login is created automatically by the system when a new extension is created.

- The user could log in the web UI with the extension number and password to access user information, extension configuration and CDR of that extension.

CREATE NEW WEB UI USER

When logged in as Super Admin, click on to create a new account for web UI user. The following dialog will prompt. Configure the parameters as shown in below table.


Figure 13: Create New User

User Name

User Password

Table 6: User Management - Create New User

Configure a username to identify the user which will be required in web UI login.

Letters, digits and underscore are allowed in the user name.

Configure a password for this user which will be required in web UI login. Letters, digits and underscore are allowed.

This is the role of the web UI user. Currently only “Admin” is 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 14: User Management – New Users


Page 47 of 304

USER PORTAL

The user could log in web UI user portal using the extension number and password. When there is an extension created in the UCM6100, the corresponding user account for the extension is automatically created. The user portal allows limited access including user information, extension configuration and CDR information of the extension. The login username is the extension number and the password is 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.

Figure 15: Edit User Information by Super Admin

The following figure shows an example of login page using extension number 1000 as the username.

Figure 16: User Portal Login

After login, the web UI displays is shown as below.


Figure 17: User Portal Layout

For the configuration parameter information in each page, please refer to

[Table 6: User Management -

Create New User]

for options in User Portal->Basic Information->User Information page;; please refer to

[EXTENSIONS]

for options in User Portal->Basic Information->Extension page;; please refer to

[CDR]

for

User Portal->Basic Information->CDR page.

CONCURRENT MULTI-‐‑USER LOGIN

When there are multiple web UI users created, concurrent multi-user login is supported on the UCM6100.

Multiple users could edit options and have configurations take effect simultaneously. However, if different users are editing the same option or making the same operation (by clicking on “Apply Changes”), a prompt will pop up as shown in the following figure.

Figure 18: Multiple User Operation Error Prompt

OPERATION LOG

Super Admin has the authority to view operation logs on UCM6100 web GUI->Settings->User

Management->Operation Log page. Operation logs list operations done by all the web UI users, for example, web UI login, creating trunk, creating outbound rule and etc. There are 6 columns to record the

operation details “Date”, “User Name”, “IP Address”, “Results”, “Page Operation” and “Specific Operation”.


Figure 19: 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.

Table 7: Operation Log Column Header

Date

User Name

IP Address

Results

Page Operation

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

Click on to view the options and values configured by this operation.

User could also filter the operation logs by time condition, IP address and/or username. Configure these conditions and then click on .


Figure 20: 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

operation logs at once.

to delete all

CHANGE BINDNG EMAIL

UCM6100 allows user to configure binding email in case login password is lost. UCM6100 login credential

will be sent to the designated email address. The feature can be found under web UI->Settings->User

Management->Change Binding Email.


Figure 21: Change Binding Email

Table 8: Change Binding Email option


Page 51 of 304

Enter the password of the account

Enter the current login user credential for UCM6100

Email Address

Email Address is used to retrieve password when password is lost

NETWORK SETTINGS

After successfully connecting the UCM6100 to the network for the first time, users could login the Web GUI

and go to Settings->Network Settings to configure the network parameters for the device.

The network setting options are similar for UCM6108 and UCM6116. Additional network functions and settings are available for UCM6102 and UCM6104:

•

UCM6102 supports Route/Switch/Dual mode functions.

•

UCM6104 supports Switch/Dual mode functions.

In this section, all the available network setting options are listed for each model. Select each tab in web

GUI->Settings->Network Settings page to configure LAN settings, WAN settings (UCM6102 only),

802.1X and Port Forwarding (UCM6102 only).

BASIC SETTINGS

Please refer to the following tables for basic network configuration parameters on UCM6102, UCM6104, and UCM6108/UCM6116 respectively.

Table 9: UCM6102 Network Settings->Basic Settings

Method

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

The default setting is "Route".

•

Route

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

•

Switch

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

•

Dual

Both ports can be used for uplink connection. Users will need assign LAN 1 or

LAN 2 as the default interface in option "Default Interface" and configure


Preferred DNS Server

"Gateway IP" for this interface.

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

IP Address

Subnet Mask

Gateway IP

DNS Server 1

DNS Server 2

User Name

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.

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

Layer 2 QoS 802.1p

Priority Value

Enter the password to connect via PPPoE.

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

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

LAN (when Method is set to "Route")

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

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

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

DNS Server 1

DNS Server 2

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

Enter DNS server address 2. The default setting is 208.67.222.222.

Allow IP Address From Enter the DHCP IP Pool starting address. The default setting is 192.168.2.100.

Allow IP Address To Enter the DHCP IP Pool ending address. The default setting is 192.168.2.254.

Default IP Lease Time Enter the IP lease time (in seconds). The default setting is 43200.

LAN (when Method is set to "Switch")

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

IP Address

Subnet Mask

Gateway IP



255.255.0.0.


DNS Server 1

DNS Server 2


0.0.0.0.



User Name

Password

Layer 2 QoS

802.1Q/VLAN Tag

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

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

LAN 1 (mapped to UCM6102 WAN port) or LAN 2 (mapped to UCM6102 LAN port) and then configure network settings for LAN 1/LAN 2. The default interface is

LAN 2.

IP Method

IP Address



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

0.

Subnet Mask

Gateway IP

DNS Server 1

DNS Server 2

User Name

Password

Layer 2 QoS

802.1Q/VLAN Tag




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.


0.0.0.0.





0.

Layer 2 QoS 802.1p

Priority Value


Method

Table 10: UCM6104 Network Settings->Basic Settings

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

•

Switch

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

•

Dual

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

LAN 2 as the default interface in option "Default Interface" and configure

"Gateway IP" for this interface.


Preferred DNS Server

Enter the preferred DNS server address. If Preferred DNS is used, UCM will try to use it as Primary DNS server.

LAN (when Method is set to "Switch")

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

IP Address

Subnet Mask

Gateway IP

DNS Server 1

DNS Server 2

User Name

Password



255.255.0.0.



0.0.0.0.




Layer 2 QoS

802.1Q/VLAN Tag

Layer 2 QoS 802.1p

Priority Value


0.


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

Default Interface

IP Method

IP Address

Subnet Mask

Gateway IP

DNS Server 1

DNS Server 2

User Name

Password

Layer 2 QoS

802.1Q/VLAN Tag

Layer 2 QoS 802.1p

Priority Value

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.




255.255.0.0.



0.0.0.0.





0.


Table 11: UCM6108/UCM6116 Network Settings->Basic Settings

Preferred DNS Server Enter the preferred DNS server address. If Preferred DNS is used, UCM will try to


IP Method

IP Address

Subnet Mask

Gateway IP

DNS Server 1

DNS Server 2

User Name

Password

Layer 2 QoS

802.1Q/VLAN Tag use it as Primary DNS server.


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.






0.

Layer 2 QoS 802.1p

Priority Value


•

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 22: UCM6102 Network Interface Method: Route

•

Method: Switch

For UCM6102, WAN port interface is used for uplink connection;; LAN port interface is used as bridge for PC connection. For UCM6104, LAN 1 port interface is used for connection;; LAN 2 port interface is used as bridge for PC connection.


Figure 23: 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.


Figure 24: UCM6102 Network Interface Method: Dual

802.1X

IEEE 802.1X is an IEEE standard for port-based network access control. It provides an authentication mechanism to device before the device is allowed to access Internet or other LAN resources. The

UCM6100 supports 802.1X as a supplicant/client to be authenticated. The following diagram and figure show UCM6100 uses 802.1X mode “EAP-MD5” on WAN port as client in the network to access Internet.


Figure 25: UCM6100 Using 802.1X as Client


Page 59 of 304

Figure 26: 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.

Table 12: UCM6100 Network Settings->802.1X

802.1X Mode

Identity

MD5 Password

802.1X Certificate

802.1X Client

Certificate

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

•

EAP-MD5

•

EAP-TLS

•

EAP-PEAPv0/MSCHAPv2

Enter 802.1X mode identity information.

Enter 802.1X mode MD5 password information.

Select 802.1X certificate from local PC and then upload.

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

STATIC ROUTES

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

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


•

Click on in the table below.

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

•

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

•

Select to delete the static route.

Table 13: UCM6100 Network Settings->Static Routes

Destination

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

UCM6100 to reach using the static route.

Example:

IP address - 192.168.66.4

IP subnet - 192.168.66.0

Netmask

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

Example:

255.255.255.0

Gateway

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

Example:

192.168.40.5

Interface

Specify the network interface on the UCM6100 to reach the destination using the

static route.

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

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

For UCM6108/UCM6116, only LAN interface is available.

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

The following diagram shows a sample application of static route usage on UCM6104.


Figure 27: 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

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 28: UCM6104 Static Route Configuration


Page 62 of 304

PORT FORWORDING (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->Settings->Network Settings->Basic

Settings page, port forwarding is available for configuration.

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

Table 14: UCM6102 Network Settings->Port Forwarding

WAN Port

Specify the WAN port number or a range of WAN ports. Up to 8 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.

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 UI 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 UI->Settings->Network Settings->Port Forwarding, configure a port forwarding entry as the figure shows below.

WAN Port: This is the port opened up on the WAN side for access purpose.

LAN IP: This is the GXP2160 IP address, under the LAN interface network of the UCM6102.

Protocol Type: We select TCP here for web UI access using HTTP.

Figure 29: UCM6102 Port Forwarding Configuration

This will allow users to access the GXP2160 web UI from public side, by typing in address “1.1.1.1:8088”.


Figure 30: GXP2160 Web Access Using UCM6102 Port Forwarding


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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 31: Register Domain Name on noip.com

2. On web UI->Settings->Network Settings->DDNS Settings, enable DDNS service and configure username, password and host name.


Figure 32: UCM6100 DDNS Setting

3. Now you can use domain name instead of IP address to connect to the UCM6100 web UI.

Figure 33: Using Domain Name to Connect to UCM6100

FIREWALL

The UCM6100 provides users firewall configurations to prevent certain malicious attack to the UCM6100 system. Users could configure to allow, restrict or reject specific traffic through the device for security and

bandwidth purpose. The UCM6100 also provides Fail2ban feature for authentication errors in SIP

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

UI->Settings->Firewall page.

STATIC DEFENSE

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


•

Current service information with port, process and type.

•

Typical firewall settings.

•

Custom firewall settings.

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

Table 15: UCM6100 Firewall->Static Defense->Current Service

Port Process Type

Protocol or Service

7777

389

22

80

8089

69

9090

Asterisk

Slapd

Dropbear

Lighthttpd

Lighthttpd

Opentftpd

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

SIP

LDAP

SSH

HTTP

HTTPS

TFTP

SIP

6060

5060

4569

5353 zero_config

Asterisk

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

UCM6100 zero_config service

SIP

SIP

UCM6100 zero_config service

37435 Syslogd udp/IPv4 Syslog

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

Table 16: Typical Firewall Settings

Ping Defense

Enable

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

WAN (UCM6102 only) interface.

Ping-of-Death

Defense Enable

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

(UCM6102 only) interface.

Under "Custom Firewall Settings", users could create new rules to accept, reject or drop certain traffic going through the UCM6100. To create new rule, click on "Create New Rule" button and a new window will

pop up for users to specify rule options.

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


•

Action: "Accept"

•

Type "In"

•

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

•

Protocol is not UDP

Rule Name

Action

Type

Service


Figure 34: Create New Firewall Rule

Table 17: 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".


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->Settings->Network Settings->Basic 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 18: UCM6100 Firewall Dynamic Defense

Dynamic Defense

Enable

Enable dynamic defense. The default setting is disabled.

Periodical Time

Interval

Blacklist Update

Interval

Connection

Threshold

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.

Dynamic Defense

Whitelist

Configure the dynamic defense whitelist.

For example,

192.168.1.3

192.168.1.4

The following figure shows a configuration example like this:


•

If a host at IP address 192.168.40.7 initiates more than 20 TCP connections to the UCM6100 within 1 minute, it will be added into UCM6100 blacklist.

•

This host 192.168.40.7 will be blocked by the UCM6100 for 300 seconds.

•

Since IP address 192.168.40.5 is in whitelist, if the host at IP address 192.168.40.5 initiates more than

20 TCP connections to the UCM6100 within 1 minute, it will not be added into UCM6100 blacklist. It can still establish TCP connection with the UCM6100.

Figure 35: Configure Dynamic Defense

FAIL2BAN

Fail2Ban feature on the UCM6100 provides intrusion detection and prevention for authentication errors in

SIP REGISTER, INVITE and SUBSCRIBE. Once the entry is detected within "Max Retry Duration", the

UCM6100 will take action to forbid the host for certain period as defined in "Banned Duration". This feature helps prevent SIP brute force attacks to the PBX system.

Table 19: Fail2Ban Settings

Global Settings

Enable Fail2Ban

Banned Duration

Max Retry Duration

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

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

Configure the duration (in seconds) for the detected host to be banned. The default setting is 300. If set to -1, the host will be always banned.

Within this duration (in seconds), if a host exceeds the max times of retry as defined in "MaxRetry", the host will be banned. The default setting is 5.


MaxRetry

Fail2Ban Whitelist

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

Configure IP address, CIDR mask or DNS host in the whiltelist. Fail2Ban will not ban the host with matching address in this list. Up to 5 addresses can be added into the list.

Local Settings

Asterisk Service

Protocol

MaxRetry

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

Fail2Ban for SIP authentication on the UCM6100.

Configure the listening port number for the service. Currently only 5060 (for UDP) is supported.

Configure the number of authentication failures during "Max Retry Duration" before the host is banned. The default setting is 10. Please make sure this option is properly configured as it will override the "MaxRetry" value under "Global

Settings".

CHANGE PASSWORD

After logging in the UCM6100 web UI 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 UI access password.

1. Go to Web UI->Settings->Change Password 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. Click on "Save" and the user will be automatically logged out.

5. Once the web page comes back to the login page again, enter the username "admin" and the new password to login.

LDAP SERVER

The UCM6100 has an embedded LDAP server for users to manage corporate phonebook in a centralized

manner.

•

By default, the LDAP server has generated the first phonebook with PBX DN

"ou=pbx,dc=pbx,dc=com" based on the UCM6100 user extensions already.


•

Users could add new phonebook with a different Phonebook DN for other external contacts. For example, "ou=people,dc=pbx,dc=com".

•

All the phonebooks in the UCM6100 LDAP server have the same Base DN "dc=pbx,dc=com".

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 other LDAP server.

To access LDAP server and client settings, go to Web GUI->Settings->LDAP Server.

LDAP SERVER CONFIGURATIONS

The following figure shows the default LDAP server configurations on the UCM6100.


Figure 36: LDAP Server Configurations


Page 72 of 304

The UCM6100 LDAP server supports anonymous access (read-only) by default. Therefore the LDAP client doesn't have to configure username and password to access the phonebook directory. The "Root DN" and

"Root Password" here are for LDAP management and configuration where users will need provide for authentication purpose before modifying the LDAP information.

The default phonebook list in this LDAP server can be viewed and edited by clicking on for the first phonebook under LDAP Phonebook.

Figure 37: Default LDAP Phonebook DN

Figure 38: Default LDAP Phonebook Attributes

LDAP PHONEBOOK

Users could use the default phonebook, edit the default phonebook, add new phonebook, import phonebook on the LDAP server as well as export phonebook from the LDAP server. The first phonebook with default phonebook dn "ou=pbx,dc=pbx,dc=com" displayed on the LDAP server page is for extensions in this PBX. Users cannot add or delete contacts directly. The contacts information will need to be modified via Web GUI->PBX->Basic/Call Routes->Extensions first. The default LDAP phonebook will then be updated automatically.


Figure 39: LDAP Server: LDAP Phonebook

•

Add new phonebook

A new sibling phonebook of the default PBX phonebook can be added by clicking on "Add" under

"LDAP Phonebook" section.

Figure 40: 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 41: 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 42: Import Phonebook

The file to be imported must be a CSV file with UTF-8 encoding. Users can open the CSV file with

Notepad and save it with UTF-8 encoding.

Here is how a sample file looks like. Please note “Account Number” and “Phonebook DN” fields are required. Users could export a phonebook file from the UCM6100 LDAP phonebook section first and use it as a sample to start with.


Figure 43: 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 44: LDAP Phonebook After Import

As the default LDAP phonebook with DN “ou=pbx,dc=pbx,dc=com” cannot be edited or deleted in

LDAP phonebook section, users cannot import contacts with Phonebook DN field “pbx” if existed in the

CSV file.

•

Export phonebook to your computer from UCM6100 LDAP server

Select the checkbox for the LDAP phonebook and then click on “Export Selected Phonebook” to export the selected phonebook. The exported phonebook can be used as a record or a sample CSV file for the users to add more contacts in it and import to the UCM6100 again.


Figure 45: Export Selected LDAP Phonebook

LDAP CLIENT CONFIGURATIONS

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

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

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 configurations for UCM6100 acting as a LDAP client.


Figure 46: LDAP Client Configurations


Page 77 of 304

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

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

Base DN: dc=pbx,dc=com

User Name: Please leave this field empty

Password: Please leave this field empty

LDAP Name Attribute: CallerIDName Email Department FirstName LastName

LDAP Number Attribute: AccountNumber MobileNumber HomeNumber Fax

LDAP Number Filter: (AccountNumber=%)

LDAP Name Filter: (CallerIDName=%)

LDAP Display Name: AccountNumber CallerIDName

LDAP Version: If existed, please select LDAP Version 3

Port: 389

The following figure shows the configuration information on a Grandstream GXP2200 to successfully use the LDAP server as configured in

Figure 36: LDAP Server Configurations

.


Figure 47: GXP2200 LDAP Phonebook Configuration


Page 78 of 304

HTTP SERVER

The UCM6100 embedded web server responds to HTTP/HTTPS GET/POST requests. Embedded HTML pages allow the users to configure the PBX through a Web browser such as Microsoft IE, Mozilla Firefox and Google Chrome. By default, the PBX can be accessed via HTTPS using Port 8089 (e.g., https://192.168.40.50:8089). Users could also change the access protocol and port as preferred under

Web GUI->Settings->HTTP Server.

Table 20: HTTP Server Settings

Redirect From Port 80

Protocol Type

Port

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

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

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

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

EMAIL SETTINGS

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

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

GUI->Settings->Email Settings.

Table 21: Email Settings

TLS Enable

Type

Domain

Enable or disable TLS during transferring/submitting your Email to other

SMTP server. The default setting is "Yes".

Select Email type.

•

MTA: Mail Transfer Agent. The Email will be sent from the configured domain. When MTA is selected, there is no need to set up SMTP server for it or no user login is required. However, the Emails sent from MTA might be considered as spam by the target SMTP server.

•

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

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


Server

Username

Password

Specify the SMTP server when using type "Client".

Username is required when using type "Client". Normally it's the Email address.

Password to login for the above Username (Email address) is required when using type "Client".

Display Name

Sender

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

Specify the sender's Email address.

For example, [email protected].

The following figure shows a sample Email settings on the UCM6100, assuming the Email is using

smtp.gmail.com as the SMTP server.

Figure 48: UCM6100 Email Settings

Once the configuration is finished, click on "Test". In the prompt, fill in a valid Email address to send a test

Email to verify the Email settings on the UCM6100.

TIME SETTINGS

AUTO TIME UPDATING

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

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


Settings->Time Auto Updating.

--------------------------------------------------------------------------------------------------------------------------------------------

Note:

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

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

--------------------------------------------------------------------------------------------------------------------------------------------

Table 22: 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

If set to "Yes", the UCM6100 is allowed to get provisioned for Time Zone from DHCP Option 2 in the local server automatically. The default setting is "Yes".

Enable DHCP Option 42

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".

Time Zone

Self-Defined 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.

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 UI->Settings->Time Settings->Set Time

Manually. The format is YYYY-MM-DD HH:MI:SS.

Figure 49: Set Time Manually

--------------------------------------------------------------------------------------------------------------------------------------------

Note:

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

NTP Server", "Enable DHCP Option 2" and "Enable DHCP Option 42" options under Web

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

--------------------------------------------------------------------------------------------------------------------------------------------


OFFICE TIME

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

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

Figure 50: Create New Office Time

Start Time

End Time

Week

Show Advanced Options

Month

Day


Table 23: Create New Office Time

Configure the start time for office hour.

Configure the end time for office hour

Select the work days in one week.

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

Select the months for office time.

Select the work days in one month.


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

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

Figure 51: Settings->Time Settings->Office Time

•

Click on to edit the office time.

•

Click on to delete the office time.

•

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

HOLIDAY

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

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


Figure 52: Create New Holiday

Table 24: Create New Holiday

Name

Holiday Memo

Month

Specify the holiday name to identify this holiday.

Create a note for the holiday.

Select the month for the holiday.

Day

Show Advanced Options

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.


Figure 53: Settings->Time Settings->Holiday

•

Click on to edit the holiday.

•

Click on to delete the holiday.

•

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

--------------------------------------------------------------------------------------------------------------------------------------------

Note:

For more details on how to use office time and holiday, please refer to the link below: http://www.grandstream.com/products/ucm_series/ucm61xx/documents/how_to_use_office_time_and_hol iday_ucm6100.pdf

--------------------------------------------------------------------------------------------------------------------------------------------

NTP SERVER

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

GUI->Settings->NTP Server. On the client side, point the NTP server address to the UCM6100 IP

address or host name to use the UCM6100 as the NTP server.


RECORDINGS STORAGE

The UCM6100 supports call recordings automatically or manually and the recording files can be saved in external storage plugged in the UCM6100 or on the UCM6100 locally. To manage the recording storage,

users can go to UCM6100 web GUI->Settings->Recordings Storage page and select whether to store the recording files in USB Disk, SD card or locally on the UCM6100.

Figure 54: Settings->Recordings Storage

•

If “Enable Auto Change” is selected, the recording files will be automatically saved in the available

USB Disk or SD card plugged into the UCM6100. If both USB Disk and SD card are plugged in, the

recording files will be always saved in the USB Disk.

•

If “Local” is selected, the recordings will be stored in UCM6100 internal storage.


•

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 55: Recordings Storage Prompt Information

Click on “OK” to continue. The users will be prompted a new dialog to select the categories for the files to be copied over.

Figure 56: 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.

LOGIN TIMEOUT SETTINGS

After the user logs in the UCM6100 web UI, the user will be automatically logged out after certain timeout.

This timeout value can be specified under UCM100 web GUI->Settings->Login Timeout Settings page.

The “User Login Timeout” value is in minute and the default setting is 10 minutes. If the user doesn’t make any operation on web UI within the timeout, the user will be logged out automatically. After that, the web UI

will be redirected to the login page and the user will need to enter username and password to log in.

If set to 0, there is no timeout for the web UI login session and the user will not be automatically logged out.

Figure 57: Login Timeout Settings



PROVISIONING

OVERVIEW

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

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

and start using the SIP devices in a managed way.

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

UI->PBX->Zero Config.

CONFIGURATION ARCHITECTURE FOR END POINT DEVICE

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

•

Global

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

Grandstream SIP devices during provisioning via Zero config.

•

Model

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

•

Device

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

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

[GLOBAL CONFIGURATION]

,

[MODEL CONFIGURATION]

and

[DEVICE

CONFIGURATION]

.


Figure 58: 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.


Figure 59: UCM6100 Zero Config

•

SIP SUBSCRIBE

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

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

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

•

DHCP OPTION 66

This method should be used on the UCM6102 because only the UCM6102 has WAN and LAN port with

LAN port supporting the router function. When the phone restarts (by default DHCP Option 66 is turned on), it will send out a DHCP DISCOVER request. The UCM6102 receives it and returns DHCP OFFER with the config server path URL in Option 66, for example, https://192.168.2.1:8089/zccgi/. The phone

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

•

mDNS

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

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


Enable Zero Config

Enable Automatic

Configuration Assignment

Automatically Assign Extension

Zero Config Extension

Segment

Enable Pick Extension

Figure 60: Auto Provision Settings

Table 25: 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.

Click on the link "Zero Config Extension Segment" to specify the extension range to be assigned if "Automatically Assign Extension" is enabled. The default range is 5000-6299. Zero Config Extension

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

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

Extensions".

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


the device's request. This feature is for the GXP series phones that support selecting extension to be provisioned via phone's LCD. The default setting is disabled.

Pick Extension Segment

Pick Extension Period (hour):

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

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

Options->General page->Extension Preference section: "Pick

Extensions".

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

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

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

configuration right away.

DISCOVERY

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

•

PING

•

ARP

•

SIP Message (NOTIFY)

Click on "Auto Discover" under web UI-> PBX-> Zero Config->Zero 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, 192.168.40.255) instead of a specific IP address. Then click on "Save" to start discovering the devices within the same network. To successfully discover the devices, "Zero Config" needs to be enabled on the UCM6100 web

GUI->PBX->Zero Config->Auto Provisioning Settings.


Figure 61: Auto Discover


Page 95 of 304

The following figure shows a list of discovered phones. The MAC address, IP Address, Extension (if assigned), Version, Vendor, Model, Connection Status, Create Config, Options (Edit/Delete/Update) are

displayed in the list.

Figure 62: Discovered Devices

GLOBAL CONFIGURATION

GLOBAL POLICY

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

•

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

•

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

•

Global Templates configuration has higher priority to Global Policy configuration.

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

•

Localization: configure display language, data and time.

•

Phone Settings: configure dial plan, call features, NAT, call progress tones and etc.

•

Contact List: configure LDAP and XML phonebook download.

•

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

•

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

•

Customization: customize LCD screen wallpaper for the supported models.

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


Figure 63: Global Policy Categories

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

Table 26: Global Policy Parameters: Localization

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.

Table 27: Global Policy Parameters: Phone Settings

Default Call Settings

Dial Plan

Enable Call Features

Use # as Dial Key

Auto Answer by Call-info

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


NAT Traversal

Use Random Port

General Settings

Call Progress Tones

HEADSET Key Mode


answer incoming calls after a short reminding beep, based on the SIP

Call-Info header sent from the server/proxy.

The default setting is enabled.

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).

•

“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 to be provisioned for more details.

Table 28: Global Policy Parameters: Contact List


LDAP Phonebook

Source

Address

Port

Base DN

User Name

Password

Number Filter

Name Filter

Version

Name Attribute

Number Attribute

Display Name

Max Hits

Search Timeout


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 UI->Settings->LDAP Server will be applied.

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

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

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

• dc=grandstream, dc=com

• ou=Boston, dc=grandstream, dc=com

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

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

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

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

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

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

LDAP search result. Example:

• gn

• cn sn description

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

LDAP search result. Example:

• telephoneNumber

• telephoneNumber Mobile

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

•

%cn %sn %telephoneNumber

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

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


Sort Results

Incoming Calls

Outgoing Calls

Lookup Display Name

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

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

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

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

XML Phonebook

Phonebook XML Server

Phonebook Download Interval

Remove manually-edited entries on download

Select the source of the phonebook XML server.

•

Disable

Disable phonebook XML downloading.

•

Manual

Once selected, users need specify downloading protocol HTTP,

HTTPS or TFTP and the server path to download the phonebook

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

•

Local UCM Server

Once selected, click on the Server Path field to upload the phonebook

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

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

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

Table 29: Global Policy Parameters: Maintenance

Upgrade and Provision

Firmware Source

Firmware source via ZeroConfig provisionoing could a URL for external server address, local UCM directory or USB media if plugged in to the

UCM6100.

Select a source to get the firmware file:

•

URL

If select to use URL to upgrade, complete the configuration for the following four parameters: “Upgrade Via”, “Server Path”, “File Prefix” and “File Postfix”.

•

Local UCM Server


Upgrade via

Server Path

File Prefix

File Postfix

Allow DHCP Option 43/66

Automatic Upgrade

Firmware Upgrade Rule

Web Access

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 as firmware source.

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

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.


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

Basic Settings

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 30: Global Policy Parameters: Network Settings

IP Address

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

PPPoE can be selected.

•

DHCP

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

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

•

PPPoE

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

Service Name for PPPoE.

Advanced Setting

Layer 3 QoS

Layer 2 QoS Tag

Layer 2 QoS Priority Value

STUN Server

Keep Alive Interval

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

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

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

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

Configure the IP address or Domain name of the STUN server. Only non-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.

Table 31: Global Policy Parameters: Customization


Wallpaper

Screen Resolution 1024 x 600




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.


•

Source


•

File




•

Source


•

File



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

•

Source


•

File



GLOBAL TEMPLATES


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

[MANAGE DEVICES]

for more details on

using the global templates.

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

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

Table 32: Create New Template

Template Name

Description

Active

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.


Figure 64: 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 UI->PBX->Zero Config->Global Templates page.

Users can click on to delete the global template or click on “Delete Selected Templates” to delete

multiple selected templates at once.


•

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

MODEL CONFIGURATION

Active

MODEL TEMPLATES

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

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

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

[MANAGE DEVICES]

for more details on using the model template.

For each created model template, users can assign it as default model template. If assigned as default model template, the values in this model template will be applied to all the devices of this model. There is

always only one default model template that can be assigned at one time on the UCM6100.

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

•

Click on “Create New Template” to add a model template.

Table 33: Create New Model Template

Model

Template Name

Description

Default Model Template

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 http://www.grandstream.com/support/tools .


Figure 65: Edit Model Template


Page 107 of 304

•

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

Config->Model Templates page.

•

Click on to delete the model template or click on “Delete Selected Templates” to delete multiple

selected templates at once.

•

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

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

UI->PBX->Zero Config->Model 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 66: 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 UI. Please contact Grandstream customer support if

the model package is needed for manual uploading.


Figure 67: Upload Model Template Manually

DEVICE CONFIGURATION

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

CREATE NEW DEVICE

Besides configuring the device after the device is discovered, users could also directly create a new device and configure basic settings before the device is discovered by the UCM6100. Once the device is plugged in, it can then be discovered and provisioned. This gives the system administrator adequate time to set up

each device beforehand.

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

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

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

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


Figure 68: Create New Device

MANAGE DEVICES

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

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

model and etc.

Figure 69: Manage Devices

to access the web UI of the phone.

•

Click on

•

Click on to edit the device configuration.

A new dialog will be displayed for the users to configure “Basic” settings and “Advanced” settings.

“Basic” settings have the same configurations as displayed when manually creating a new device, i.e., account, line key and MPK settings;; “Advanced” settings allow users to configure more details in a five-level structure.


Figure 70: Edit Device

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

(1) Global Policy

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

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

PBX->Zero Config->Global Policy.

(2) Global Templates

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

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


(3) Default Model Template

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

Please see default model template option in

[Table 33: Create New Model Template].

(4) Model Templates

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

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

(5) Customize Device Settings

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

Figure 71: Edit Customize Device Settings

Scroll down in the dialog to view and edit the device-specific options. If the users would like to add more options which are not in the pre-defined list, click on “Add New Field” to add a P value number and the value to the configuration. The following figure shows setting P value “P1362” to


“en”, which means the display language on the LCD is set to English. The warning information on right tells that the option matching the P value number exists and clicking on it will lead to the matching option. For P value information of different models, please refer to configuration template here http://www.grandstream.com/support/tools .

Figure 72: 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.


Figure 73: 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.


Figure 74: Modify Selected Devices - Different Models

--------------------------------------------------------------------------------------------------------------------------------------------

Note:

Performing batch operation will override all the existing device configuration on the page.

--------------------------------------------------------------------------------------------------------------------------------------------

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

Config page. Users could then click on to send NOTIFY to the SIP end point device and trigger the

provisioning process. The device will start downloading the generated configuration file from the URL contained in the NOTIFY message.


Figure 75: Device List in Zero Config

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

SAMPLE APPLICATION

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

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

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

1. Go to web GUI->PBX->Zero Config->Zero Config Settings, select “Enable Zero Config”.

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

Firmware Source as follows.


Figure 76: Zero Config Sample - Global Policy

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

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

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

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

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

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

“Create New Device” function to add the devices to the device list on web UI->PBX->Zero

Config->Zero Config.

6. On web GUI->PBX->Zero Config->Zero Config page, users could identify the devices by their MAC addresses or IP addresses displayed on the list. Click on to edit the device settings.

7. For each of the 5 phones used by English speaking customer support, in “Basic” settings select an available extension for account 1 and click on “Save”. Then click on “Advanced” settings tab to bring


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 77: 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.



Select “Spanish Support Template” in

○

“

Model Template”. The preview of the device settings is displayed on the right side and we can see the language is set to “Español” since Model Template has

the higher priority for the option “Language”, which overrides the value configured in default model template.

9. For the GXV3275 used by the customer support supervisor, select an available extension for account

1 on “Basic” settings and click on “Save”. Users can see the preview of the device configuration in

“Advanced” settings. There is no model template configured for GXV3275.



10. Click on “Apply Changes” to apply saved changes.

11. On the web UI->PBX->Zero Config->Zero Config page, click on to send NOTIFY to trigger the device to download config file from UCM6100.

Now all the 9 phones in the network will be provisioned with an unique extension registered on the

UCM6100. 3 of the phones will be provisioned to display Spanish on LCD and the other 5 will be provisioned to display English on LCD. The GXV3275 used by the supervisor will be provisioned to use the default language on LCD display since it’s not specified in the global policy.


EXTENSIONS

CREATE NEW USER

CREATE NEW SIP EXTENSION

To manually create new SIP user, go to Web GUI->PBX->Basic/Call Routes->Extensions. Click on

"Create New User"->"Create New SIP Extension" and a new dialog window will show for users to fill in the extension information.

Figure 80: Create New Device

SIP extension options are divided into four categories:

•

Basic Settings

•

Media

•

Features

•

Specific Time

Click on the tag to view or edit options belonging to that category.

The configuration parameters are as follows.

Table 34: SIP Extension Configuration Parameters - Basic Settings


General

Extension

CallerID Number

Permission

SIP/IAX Password

Support Hot-Desking Mode

Auth ID

Enable Voicemail

Voicemail Password

Skip Voicemail Password

Verification

Disable This Extension

Enable LDAP

User Settings

First Name

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.

If enabled, SIP Password will accept only alphabet characters and digits;;

AuthID will be changed to the same as Extension.

Configure the authentication ID for the user. If not configured, the extension number will be used for authentication.

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

Configure voicemail password (digits only) for the user to access the voicemail 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.

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.

If enabled, this extension will be added to LDAP Phonebook PBX list;; if disabled, this extension will be skipped when creating LDAP Phonebook.

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


Last Name

Email Address

User Password

Language

Music On Hold

SIP Settings

NAT

Can Reinvite

DTMF Mode

Insecure


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->Internal Options->Language. The dropdown list shows all the current available voice prompt languages on the UCM6100.

To add more languages in the list, please download voice prompt package by selecting "Check Prompt List" under web UI->PBX->Internal

Options->Language.

Select which Music On Hold class to suggest to extension when putting the active call on hold.

Table 35: SIP Extension Configuration Parameters - Media

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

By default, the UCM6100 will route the media steams from SIP endpoints through itself. If enabled, the PBX will attempt to negotiate with the endpoints to route the media stream directly. It is not always possible for the UCM6100 to negotiate endpoint-to-endpoint media routing. The default setting is "No".

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

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

"Inband" is selected, 64-kbit PCMU and PCMA are required. When "Auto" is selected, RFC2833 will be used if offered, otherwise "Inband" will be used.

•

Port: Allow peers matching by IP address without matching port number.

•

Very: Allow peers matching by IP address without matching port number. Also, authentication of incoming INVITE messages is not required.

•

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


Enable Keep-alive

Keep-alive Frequency

TEL URI

SRTP

Fax Mode

Strategy

INVITE.

The default setting is "Port".

If enabled, empty SDP packet will be sent to the SIP server periodically to keep the NAT port open. The default setting is "Yes".

Configure the Keep-alive interval (in seconds) to check if the host is up.

The default setting is 60 seconds.

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.

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

This option controls how the extension can be used on devices within different types of network.

•

Allow All

Device in any network can register this extension.

•

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.

Codec Preference

Call Transfer

Call Forward Unconditional

CFU Time Condition

Table 36: SIP Extension Configuration Parameters - Features

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


Call Forward No Answer

CFN Time Condition

Call Forward Busy

CFB Time Condition

CC Settings

Enable CC


only during the selected time condition. The available time condition are

“Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of

Office Time or Holiday” and “Specific”.

Note:

•

“Specific” has higher priority to “Office Times” if there is a conflict in terms of time period.

•

Specific time can be configured on the bottom of the extension configuration dialog. Scroll down the add Time Condition for specific time.

•

Office Time and Holiday could be configured on page Settings->Time

Settings->Office Time/Holiday page.

Configure the Call Forward No Answer target number. If not configured, the Call Forward No Answer feature is deactivated. The default setting is deactivated.

Select time condition for Call Forward No Answer. The available time condition are “Office Time”, “Out of Office Time”, “Holiday”, “Out of

Holiday”, “Out of Office Time or Holiday” and “Specific”.

Note:

•


•


•



Configure the Call Forward Busy target number. If not configured, the Call

Forward Busy feature is deactivated. The default setting is deactivated.

Select time condition for Call Forward Busy. The available time condition are “Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of


Note:

•


•


•



If enabled, UCM6100 will automatically alert this extension when a called


CC Mode

CC Max Agents

CC Max Monitors

Ring Simultaneously

Ring Simultaneously

External Number

Time Condition for Ring

Simultaneously

Other Settings

Ring Timeout

Auto Record

Skip Trunk Auth


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.

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. 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.

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.

Configure the number of seconds to ring the user before the call is forwarded to voicemail (voicemail is enabled) or hang up (voicemail is disabled). If not specified, the default ring timeout is 60 seconds on the

UCM6100, which can be configured in the global ring timeout setting under web GUI->Internal Options->IVR Prompt: General Preference. The valid range is between 5 seconds and 600 seconds.

Note:

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

Enable automatic recording for the calls using this extension. The default setting is disabled. The recording files can be accessed under web

GUI->CDR->Recording Files.

· 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.

Time Condition for Skip Trunk

Auth

Dial Trunk Password

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.

Table 37: SIP Extension Configuration Parameters - Specific Time

Specific Time

Time Condition

Click to add Time Condition to configure specific time for this extension.

CREATE NEW IAX EXTENSION

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->PBX->Basic/Call Routes->Extensions. Click on

"Create New User"->"Create New IAX Extension" and a new dialog window will show for users to fill in the extension information. The configuration parameters are as follows.

Table 38: IAX Extension Configuration Parameters - Basic Settings

General

Extension

CallerID Number

Permission

SIP/IAX Password



Note:


VoIP provider.



Note:


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


Enable Voicemail

Voicemail Password


Verification


Enable LDAP

User Settings

First Name

Last Name

Email Address

User Password

Language

Music On Hold






Note:






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 which Music On Hold class to suggest to the extension when putting the active on hold.

Table 39: IAX Extension Configuration Parameters - Media

SIP Settings


Max Number of Calls

Require Call Token

SRTP

Fax Mode

Strategy

Codec Preference

Call Transfer


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

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

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

Select Fax Mode. The default setting is “None”.

•

None: Disable Fax. This is the default setting.

•




•

Allow All


•

Local Subnet Only


•

A Specific IP Address




PCMU, PCMA, GSM, AAL2-G.726-32, G,726, G.722, G.729, G.723, iLBC, ADPCM, H.264, H.263 and H.263p.

Table 40: IAX Extension Configuration Parameters - Features

CFU Time Condition


Select time condition for Call Forward Unconditional. CFU takes effect only during the selected time condition. The available time condition are



Note:

•

“Specific” has higher priority to “Office Times” if there is a conflict in



CFN Time Condition

Call Forward Busy

CFB Time Condition


Ring Simultaneously

External Number


terms of time period.

•


•






Note:

•


•


•







Note:

•


•


•



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.




Simultaneously

Other Settings


Ring Timeout

Auto Record

Skip Trunk Auth


Auth

Dial Trunk Password


UCM6100, which can be configured in the global ring timeout setting under web GUI->Internal Options->IVR Prompt: General Preference. The valid range is between 5 seconds and 600 seconds.

Note:




•

If 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.


Table 41: IAX Extension Configuration Parameters - Specific Time

Specific Time

Time Condition Click to add Time Condition to configure specific time for this extension.

CREATE NEW FXS EXTENSION

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->PBX->Basic/Call Routes->Extensions. Click on

"Create New User"->"Create New FXS Extension" and a new dialog window will show for users to fill in the extension information. The configuration parameters are as follows.


General

Extension

Analog Station

CallerID Number

Permission

Enable Voicemail

Voicemail Password


Verification


Enable LDAP

User Settings

First Name

Last Name

Email Address

User Password

Table 42: FXS Extension Configuration Parameters - Basic Settings


Select the FXS port to be assigned for this extension.


Note:


VoIP provider.



Note:







Note:






Configure the password for user portal access. A random numeric


Language

Music On Hold

Analog Settings

Call Waiting

User ‘#’ as SEND

RX Gain

TX Gain

MIN RX Flash

MAX RX Flash

Enable Polarity Reversal

Echo Cancellation


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




Select which Music On Hold class to suggest to extension when putting the active call on hold.

Table 43: FXS Extension Configuration Parameters - Media

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

"No".

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

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

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

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

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

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

If enabled, a polarity reversal will be marked as received when an outgoing call is answered by the remote party. For some countries, a polarity reversal is used for signaling the disconnection of a phone line and the call will be considered as hangup on a polarity reversal. The default setting is "Yes".

Specify "ON", "OFF" or a value (the power of 2) from 32 to 1024 as the number of taps of cancellation.

Note:

When configuring the number of taps, the number 256 is not translated into 256ms of echo cancellation. Instead, 256 taps means 256/8 = 32 ms.


3-Way Calling

Send CallerID After

Fax Mode

Call Transfer


CFU Time Condition


CFN Time Condition

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”.

•

None: Disable Fax.

•



•

Fax Gateway: If selected, the UCM6100 can support conversation and processing of Fax data from T.30 to T.38 or T.38 to T.30. This feature is only available for FXS or FXO port.

Table 44: FXS Extension Configuration Parameters - Features


Select time condition for Call Forward Unconditional. CFU takes effect only during the selected time condition. The available time condition are



Note:

•


•


•







Call Forward Busy

CFB Time Condition

CC Settings

Enable CC


Ring Simultaneously

External Number


Simultaneously

Other Settings

Ring Timeout


Note:

•


•


•







Note:

•


•


•



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.

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.




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


Auto Record

Skip Trunk Auth


Auth

Dial Trunk Password under web GUI->Internal Options->IVR Prompt: General Preference. The valid range is between 5 seconds and 600 seconds.

Note:




•

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.


Table 45: FXS Extension Configuration Parameters - Specific Time

Specific Time

Time Condition

Click to add Time Condition to configure specific time for this extension.

BATCH ADD EXTENSIONS

BATCH ADD SIP EXTENSIONS

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

SIP Extensions".

Table 46: Batch Add SIP Extension Parameters

General

Start Extension

Configure the starting extension number of the batch of extensions to be added.


Create Number

Permission

Enable Voicemail

SIP/IAX Password

Voicemail Password

Ring Timeout

Auto Record


Verification

Music On Hold

Enable LDAP


Specify the number of extensions to be added. The default setting is 5.



Note:

Users need to have the same level as or higher level than an outbound rule's privilege in order to make outbound calls from this rule.

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

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

•

User Random Password.

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

•

Use Extension as Password.

•

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

Configure Voicemail password (digits only) for the users.

•


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

•


•



UCM6100, which can be configured in the global ring timeout setting under web GUI->Internal Options->IVR Prompt: General Preference.

The valid range is between 5 seconds and 600 seconds.

Note:



GUI->CDR->Recording Files.



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


SIP Settings

NAT

Can Reinvite

DTMF Mode

Insecure

TEL URI

Other Settings

SRTP

Fax Mode

Enable Keep-alive

Keep-alive Frequency


PBX list;; if disabled, the batch added extensions will be skipped when creating LDAP Phonebook.

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

By default, the PBX will route the media steams from SIP endpoints through itself. If enabled, the PBX will attempt to negotiate with the endpoints to route the media stream directly. It is not always possible for the PBX to negotiate endpoint-to-endpoint media routing. The default setting is "No".

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

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

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

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

•

Port: Allow peers matching by IP address without matching port number.

•

Very: Allow peers matching by IP address without matching port number. Also, authentication of incoming INVITE messages is not required.

•

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

INVITE.

The default setting is "Port".

If enabled, empty SDP packet will be sent to the SIP server periodically to keep the NAT port open. The default setting is "Yes".

Configure the number of seconds for the host to be up for Keep-alive. The default setting is 60 seconds.

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.

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


Strategy

Skip Trunk Auth

Codec Preference 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



•

Allow All


•

Local Subnet Only


•

A Specific IP Address.



If enable “All”, users do not need to enter password when making an outbound call. If enable “Follow Me”, the user can dial out via follow me without password.


PCMU, PCMA, GSM, AAL2-G.726-32, G.722, G.729, G.723, iLBC,

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

BATCH ADD IAX EXTENSIONS

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

IAX Extensions".

Table 47: Batch Add IAX Extension Parameters

General

Start Extension

Create Number

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.



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

Voicemail Password

Ring Timeout

Auto Record


Verification

Music On Hold

Enable LDAP

IAX Settings

Max Number of Calls

Require Call Token


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

•


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

•


•


Configure Voicemail password (digits only) for the users.

•


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

•


•



UCM6100, which can be configured in the global ring timeout setting under web GUI->Internal Options->IVR Prompt: General Preference.

The valid range is between 5 seconds and 600 seconds.

Note:






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 number of calls allowed for each remote IP address.

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


default setting is "Yes".

Other Settings

SRTP

Fax Mode

Strategy

Skip Trunk Auth

Codec Preference

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 setting page under web UI->PBX->Internal Options->Fax/T.38.


•

Allow All


•

Local Subnet Only


•

A Specific IP Address.



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.


PCMU, PCMA, GSM, AAL2-G.726-32, G.722, G.729, G.723, iLBC,

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

SEARCH AND EDIT EXTENSION

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

"Edit” , "Reboot" and "Delete" are available per extension. User can search an extension by specifying the extension number to find an extension quickly.


Figure 81: Manage Extensions

•

Status

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

Green: Free

Blue:

Yellow:

Grey:

Ringing

In Use

Unavailable (the extension is not registered or disabled on the PBX)

•

Edit single extension

Click on to start editing the extension parameters.

•

Reboot the user

Click on to send NOTIFY reboot event to the device which has an UCM6100 extension already registered. To successfully reboot the user, "Zero Config" needs to be enabled on the UCM6100 web

GUI->PBX->Zero Config->Auto Provisioning Settings.

•

Delete single extension

Click on to delete the extension. Or select the checkbox of the extension and then click on "Delete

Selected Extensions".

•

Modify selected extensions

Select the checkbox for the extension(s). Then click on "Modify Selected Extensions" to edit the

extensions in a batch.

•

Delete selected extensions

Select the checkbox for the extension(s). Then click on "Delete Selected Extensions" to delete the extension(s).


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 82: Export Extensions

The exported csv file can be serve as a template for users to fill in desired extension information to be

imported to the UCM6100.

IMPORT EXTENSIONS

The capability to import extensions to the UCM6100 provides users flexibility to batch add extensions with

similar or different configuration quickly into the PBX system.

1. Export extension csv file from the UCM6100 by clicking on "Export Extensions" button.

2. Fill up the extension information you would like in the exported csv template.

3. Click on "Import Extensions" button. The following dialog will be prompted.

Figure 83: Import Extensions

4. Select the option in "On Duplicate Extension" to define how the duplicate extension(s) in the imported

csv file should be treated by the PBX.

•

Skip: Duplicate extensions in the csv file will be skipped. The PBX will keep the current extension


information as previously configured without change.

•

Delete and Recreate: The current extension previously configured will be deleted and the duplicate extension in the csv file will be loaded to the PBX.

•

Update Information: The current extension previously configured in the PBX will be kept. However, if the duplicate extension in the csv file has different configuration for any options, it will override the configuration for those options in the extension.

5. Click on to select csv file from local directory in the PC.

6. Click on "Save" to import the csv file.

7. Click on "Apply Changes" to apply the imported file on the UCM6100.

EMAIL TO USER

Once the extensions are created with Email address, the PBX administrator can click on button “Email To

User” to send the account registration and configuration information to the user. Please make sure Email setting under web UI->Settings->Email Settings is properly configured and tested on the UCM6100

before using “Email To User”.

When click on “Email To User” button, the following message will be prompted in the web page. Click on

OK to confirm sending the account information to all users’ Email addresses.

Figure 84: Email To User - Prompt Information

The user will receive Email including account registration information and LDAP configuration. A QR code

is also generated for Mobile applications to scan it and get automatically provisioned. QR code provisioning is supported on Grandstream Softphone GS Wave Android

TM

application.


Figure 85: Account Registration Information and QR Code


Figure 86: LDAP Client Information and QR Code



TRUNKS

ANALOG TRUNKS

Go to Web GUI->PBX->Basic/Call Routes->Analog Trunks to add and edit analog trunks.

•

Click on "Create New Analog Trunk" to add a new analog trunk.

•

Click on to edit the analog trunk.

•

Click on to delete the analog trunk.

ANALOG TRUNK CONFIGURATION

The analog trunk options are listed in the table below.

Table 48: Analog Trunk Configuration Parameters

Channels

Trunk Name

SLA Mode

Barge Allowed

Hold Access

Select the channel for the analog trunk.

•

UCM6102: 2 channels

•

UCM6104: 4 channels

•

UCM6108: 8 channels

•

UCM6116: 16 channels

Specify a unique label to identify the trunk when listed in outbound rules, incoming rules and etc.

Enable this option to satisfy two primary use cases, which include emulating a simple key system and creating shared extensions on a PBX.

Enable SLA Mode will disable polarity reversal.

The barge option specifies whether or not other stations are allowed to join a call in progress on this trunk. If enabled, the other stations can press the line button to join the call. The default setting is Yes.

The 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

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 on Answer Delay

Current Disconnect Threshold

(ms)

Ring Timeout

RX Gain

TX Gain

Use CallerID

Fax Mode

Caller ID Scheme

FXO Dial Delay(ms)

Auto Record

Disable This Trunk polarity reversal is used for signaling the disconnection of a phone line and the call will be considered as “hangup” on a polarity reversal. The default setting is “No”.

When FXO port answers the call, FXS may send a Polarity Reversal. If this interval is shorter than the value of “Polarity on Answer Delay”, the

Polarity Reversal will be ignored. Otherwise, the FXO will onhook to disconnect the call. The default setting is 600ms.

This is the periodic time (in ms) that the UCM6100 will use to check on a voltage drop in the line. The default setting is 200. The valid range is 50 to

3000.

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

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

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

Configure the RX gain for the receiving channel of analog FXO port. The valid range is from -13.5 (dB) to + 12.0 (dB). The default setting is 0.

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

The valid range is from -13.5 (dB) to + 12.0 (dB). The default setting is 0.

Configure to enable CallerID detection. The default setting is “Yes”.

There are three options available in Fax Mode. The default setting is

“None”.

•

None: Disable Fax.

•



•

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.

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


If selected, the trunk will be disabled.


DAHDI Out Line Selection

Tone Settings

Congestion Detection

Congestion Count

Tone Country

Busy Detection

Busy Tone Count

Busy Tone


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's idle).

The default setting is “Ascend” mode.

Busy Detection is used to detect far end hangup or for detecting busy signal. The default setting is "Yes".

If "Busy Detection" is enabled, users can specify the number of busy tones to be played before hanging up. The default setting is 2. Better results might be achieved if set to 4, 6 or even 8. Please note that the higher the number is, the more time is needed to hangup the channel.

However, this might lower the probability to get random hangup.

Congestion detection is used to detect far end congestion signal. The default setting is "Yes".

If "Congestion Detection" is enabled, users can specify the number of congestion tones to wait for. The default setting is 2.

Select the country for tone settings. If "Custom" is selected, users could manually configure the values for Busy Tone and Congestion Tone. The default setting is "United States of America (USA)".

Syntax:

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

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].


Congestion Tone

PSTN Detection

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

Default value:

f1=480@-50,f2=620@-50,c=500/500

Syntax:

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

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

Frequencies Range: [0, 4000)

Busy Level Range: (-300, 0)

Cadence Range: [0, 16383].

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

Default value:

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

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

Disconnect by PSTN. Before the detecting, please make sure there are more than one channel configured and working properly. If the detection has busy tone, the "Tone Country" option will be set as "Custom".

PSTN DETECTION

The UCM6100 provides PSTN detection function to help users detect the busy tone, Polarity Reversal and

Current Disconnect by making a call from the PSTN line to another destination. The detecting call will be answered and up for about 1 minute. Once done, the detecting result will show and can be used for the

UCM6100 settings.

1. Go to UCM6100 web GUI->PBX->Basic/Call Routes->Analog Trunks page.

2. Click to edit the analog trunk created for the FXO port.

3. In the dialog window to edit the analog trunk, go to "Tone Settings" section and there are two methods to set the busy tone.

•

Tone Country. The default setting is "United States of America (USA)".

•

PSTN Detection.


Figure 87: UCM6100 FXO Tone Settings

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

Figure 88: UCM6100 PSTN Detection

•

If there are two FXO ports connected to PSTN lines, use the following settings for auto-detection.

Detect Model: Auto Detect.

Source Channel: The source channel to be detected.

Destination Channel: The channel to help detecting. For example, the second FXO port.

Destination Number: The number to be dialed for detecting. This number must be the actual

PSTN number for the FXO port used as the destination channel.


Figure 89: UCM6100 PSTN Detection: Auto Detect

•

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

Figure 90: UCM6100 PSTN Detection: Semi-Auto Detect

Detect Model: Semi-auto Detect.

Source Channel: The source channel to be detected.

Destination Number: The number to be dialed for detecting. This number could be a cell phone number or other PSTN number that can be reached from the source channel PSTN number.

5. Click "Detect" to start detecting. The source channel will initiate a call to the destination number. For

"Auto Detect", the call will be automatically answered. For "Semi-auto Detect", the UCM6100 web GUI

will display prompt to notify the user to answer or hang up the call to finish the detecting process.

6. Once done, the detected result will show. Users could save the detecting result as the current

UCM6100 settings.


Table 49: PSTN Detection for Analog Trunk

Detect Model

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.

Source Channel

Destination Channel

Destination Number Configure the number to be called to help the detection.

--------------------------------------------------------------------------------------------------------------------------------------------

Note:

•

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

•

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

•

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

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

configuration.

--------------------------------------------------------------------------------------------------------------------------------------------

VOIP TRUNKS

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

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

Options to edit/detect the trunk.


•

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

•

Click on to configure detailed parameters for the VoIP trunk.

•

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

•

Click on to start LDAP Sync.

•

Click on to delete the VoIP trunk.

For VoIP trunk example, please refer to the document in the following link: http://www.grandstream.com/products/ucm_series/ucm65xx/documents/how_to_interconnect_multiple_uc

m6510_using_peer_trunk.pdf

The VoIP trunk options are listed in the table below.

Table 50: Create New SIP Trunk

Type

Provider Name

Host Name

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 Original CID

Keep Trunk CID

NAT

Disable This Trunk

TEL URI

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.

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


Need Registration

Username

Password

Auth ID

Auto Record

Basic Settings

Provider Name

Host Name

Transport

Keep Original CID

Keep Trunk CID

NAT


the Request-Line and TO header in the SIP request to indicate the E.164 number. If set to "Enable", "Tel:" will be used instead of "SIP:" in the SIP request. The default setting is disabled.

Select whether the trunk needs to register on the external server or not when "Register SIP Trunk" type is selected. The default setting is No.

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

"Register SIP Trunk" type is selected.

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

"Register SIP Trunk" is selected.

Enter the Authentication ID for "Register SIP Trunk" type.

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

Table 51: SIP Register Trunk Configuration Parameters



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

•

UDP Only

•

TCP Only

•

TLS Only

•

All - UDP Primary: UDP is the primary transport protocol when all the other SIP transport methods are available too.

•

All - TCP Primary: TCP is the primary transport protocol when all the other SIP transport methods are available too.

•

All – TLS Primary: TLS is the primary transport protocol when all the other SIP transport methods are available too.

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.


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


Disable This Trunk

TEL URI

Need Registration

Username

Password

Auth ID

Auto Record

Advanced Settings

Codec Preference

From Domain

From User

Send PPI Header


related to NAT configuration or SIP/RTP port configuration on the firewall.


Note:



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.


"Register SIP Trunk" type is selected.


"Register SIP Trunk" is selected.

Enter the Authentication ID for "Register SIP Trunk" type.


Select audio and video codec for the VoIP trunk. The available codecs are: PCMU, PCMA, GSM, AAL2-G.726-32, G.726, G.722, G.729, G.723, iLBC, ADPCM, H.264, H.263, H.263p.

Configure the actual domain name where the extension comes from. This can be used to override the From Header.

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].

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


Send PAI Header

Outbound Proxy Support

Outbound Proxy

DID Mode

DTMF Mode same time. Only one of the two headers is allowed to be contained in the

SIP INVITE message.

If enabled, the SIP INVITE message sent to the trunk will contain PAI

(P-Asserted-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 is allowed to be contained in the

SIP INVITE message.

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

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

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

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

Configure the default DTMF mode when sending DTMF on this trunk.

•

Default: The global setting of DTMF mode will be used. The global setting for DTMF Mode setting is under web UI->PBX->SIP

Settings->ToS.

•

RFC2833: Send DTMF using RFC2833.

•

Info: Send DTMF using SIP INFO message.

•

Inband: Send DTMF using inband audio. This requires 64 bit codec, i.e., PCMU and PCMA.

•

Auto: Send DTMF using RFC2833 if offered. Otherwise, inband will be used.

Enable Qualify

Qualify Timeout

Qualify Frequency

Maximum Number of Call Lines

Fax Mode

If enabled, the UCM6100 will regularly send SIP OPTIONS to the device to check if the device is still online. The default setting is "No".

When "Enable Qualify" option is set to "Yes", configure the timeout (in ms) for the Qualify SIP message. If no response is received within the timeout, the device is considered offline. The default setting is 1000ms.

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

The maximum number of concurrent calls using the trunk. The default settings 0, which means no limite.


•

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


SRTP

CC Settings

Enable CC

Basic Settings

Provider Name

Host Name

CC Max Agents

CC Max Monitors

Transport

Keep Original CID

Keep Trunk CID

NAT


user, the Fax will be sent to the default Email address configured in


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

If enabled, the system will automatically alert the user when a called party is available, given that a previous call to that party failed for some reason.



Table 52: SIP Peer Trunk Configuration Parameters



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

•

UDP Only

•

TCP Only

•

TLS Only

•

All - UDP Primary: UDP is the primary transport protocol when all the other SIP transport methods are available too.

•

All - TCP Primary: TCP is the primary transport protocol when all the other SIP transport methods are available too.

•

All – TLS Primary: TLS is the primary transport protocol when all the other SIP transport methods are available too.

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.


Turn on this option when the PBX is using public IP and communicating


Disable This Trunk

TEL URI

Caller ID

CallerID Name

Auto Record


Codec Preference

DID Mode

DTMF Mode


awith 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.


Note:



If the trunk has an assigned PSTN telephone number, this field should be set to "User=Phone". Then a "User=Phone" parameter will be attached to the Request-Line and TO header in the SIP request to indicate the E.164 number. If set to "Enable", "Tel:" will be used instead of "SIP:" in the SIP request. The default setting is disabled.

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

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

•

The CallerID configured for the extension will be looked up first.

•

If no CallerID configured for the extension, the CallerID configured for the trunk will be used.

•

If the above two are missing, the "Global Outbound CID" defined in

Web GUI->PBX->Internal Options->General will be used.

Configure the name of the caller to be displayed when the extension has no CallerID Name configured.



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

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

Configure the default DTMF mode when sending DTMF on this trunk.

•

Default: The global setting of DTMF mode will be used. The global setting for DTMF Mode setting is under web UI->PBX->SIP

Settings->ToS.

•

RFC2833: Send DTMF using RFC2833.

•

Info: Send DTMF using SIP INFO message.


Enable Qualify

Qualify Timeout

Sync LDAP Password

Sync LDAP Port

LDAP Outbound Rule

LDAP Dialed Prefix

•

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.



Qualify Frequency


Fax Mode

SRTP

Sync LDAP Enable


The maximum number of concurrent calls using the trunk. The default settings 0, which means no limite.


•

None: Disable Fax.

•



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

If enabled, the local UCM6100 will automatically provide and update the local LDAP contacts to the remote UCM6100 SIP peer trunk. In order to ensure successful synchronization, the remote UCM6100 peer also needs to enable this option on the SIP peer trunk. The default setting is

"No".

This is the password used for LDAP contact file encryption and decryption during the LDAP sync process. The password must be the same on both

UCM6100 peers o ensure successful synchronization.

Configure the TCP port used LDAP sync feature between two peer

UCM6100.

Specify an outbound rule for LDAP sync feature. The UCM6100 will automatically modify the remote contacts by adding prefix parsed from this rule.

Specify the prefix for LDAP sync feature. The UCM6100 will automatically modify the remote contacts by adding this prefix.

CC Settings

Enable CC If enabled, the system will automatically alert the user when a called party


Provider Name

Host Name

Keep Trunk CID

Username

Password

Disable This Trunk

Basic Settings

Provider Name

Host Name

Keep Trunk CID

Disable This Trunk

Caller ID

CC Max Agents

CC Max Monitors

Type


is available, given that a previous call to that party failed for some reason.



Table 53: Create New IAX Trunk

Select the VoIP trunk type.

•

Peer IAX Trunk

•

Register IAX 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".


"Register IAX Trunk" type is selected.


"Register IAX Trunk" type is selected.


Table 54: IAX Register Trunk Configuration Parameters







CallerID Name


•


•


•




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

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

Username

Password


Codec Preference

Enable Qualify

Qualify Timeout

Qualify Frequency


Fax Mode





The maximum number of concurrent calls using the trunk. The default settings 0, which means no limited.


•

None: Disable Fax.

•



Table 55: IAX Peer Trunk Configuration Parameters

Basic Settings

Provider Name



Host Name

Keep Trunk CID

Disable This Trunk

Caller ID

CallerID Name






•


•


•





Codec Preference

Enable Qualify

Qualify Timeout

Qualify Frequency


Fax Mode





The maximum number of concurrent calls using the trunk. The default settings 0, which means no limited.


•

None: Disable Fax.

•




DIRECT OUTWARD DIALING (DOD)

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

Example of how DOD is used:

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

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

This poses a problem as the CEO would like their calls to come from their direct line. This can be accomplished by configuring DOD for the CEO’s extension.

Steps on how to configure DOD on the UCM6100:

1. To setup DOD go to UCM6100 web GUI->PBX->Basic/Call Routes->VoIP Trunks page.

2. Click to access the DOD options for the selected SIP Trunk.

3. Click "Create a new DOD" to begin your DOD setup

4. For "DOD Number" enter one of the numbers (DIDs) from your SIP trunk provider. In the example above Company ABC received 4 DIDs from their provider. ABC will enter in the number for the CEO's direct line.

5. Select an extension from the "Available Extensions" list. Users have the option of selecting more than one extension. In this case, Company ABC would select the CEO's extension. After making the selection, click on the button to move the extension(s) to the "Selected Extensions" list.


Page 164 of 304

Figure 91: DOD extension selection

6. Click "Save" at the bottom.

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

Figure 92: Edit DOD


SLA STATION

Starting from 1.0.7.10, the UCM6100 supports SLA that allows mapping the key with LED on a multi-line phone to different external lines. When there is an incoming call and the phone starts to ring, the LED on the key will flash in red and the call can be picked up by pressing this key. This allows users to know if the line is occupied or not. The SLA function on the UCM6100 is similar to BLF but SLA is used to monitor external line i.e., analog trunk on the UCM6100. Users could configure the phone with BLF mode on the

MPK to monitor the analog trunk status or press the line key pick up call from the analog trunk on the

UCM6100.

CREATE/EDIT SLA STATION

SLA Station can be configured on web GUI->PBX->Basic/Call Routes->SLA Station.

Figure 93: SLA Station

•

Click on “Create New SLA Station” to add a SLA Station.

•

Click on parameters.

to edit the SLA Station. The following table shows the SLA Station configuration

•

Click on to delete the SLA Station.

Station Name

Station

Available SLA Trunks

Selected SLA Trunks

Table 56: 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

Ring Delay

Hold Access

Configure the time (in seconds) to ring the station before the call is considered unanswered. No timeout is set by default. If set to 0, there will be no timeout.

Configure the time (in seconds) for delay before ringing the station when a call first coming in on the shared line. No delay is set by default. If set to

0, there will be no delay.

This option defines the competence of the hold action for one particular trunk. If set to “open”, any station could hold a call on that trunk or resume one held session;; if set to “private”, only the station that places the trunk call on hold could resume the session. The default setting is “open”.

SAMPLE CONFIGURATION

1. On the UCM6100, go to web UI->Basic/Call Routes->Analog Trunks page. Create analog trunk or edit the existing analog trunk. Make sure “SLA Mode” is enabled for the analog trunk. Once enabled,

this analog trunk will be only available for the SLA stations created under web UI->Basic/Call

Routes->SLA Station page.


Figure 94: Enable SLA Mode for Analog Trunk

Click on “Save”. The analog trunk will be listed with trunk mode “SLA”.

Figure 95: Analog Trunk with SLA Mode Enabled

2. On the UCM6100, go to web UI->Basic/Call Routes->SLA Station page, click on “Create New SLA

Station”. Please refer to section

[CREATE/EDIT SLA STATION]

for the configuration parameters. Users can create one or more SLA stations to monitor the analog trunk. The following figure shows two stations, 1002 and 1005, are configured to be associated with SLA trunk “fxo1”.

Figure 96: SLA Example - SLA Station


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 97: 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.


CALL ROUTES

OUTBOUND ROUTES

In the UCM6100, an outgoing calling rule pairs an extension pattern with a trunk used to dial the pattern.

This allows different patterns to be dialed through different trunks (e.g., "Local" 7-digit dials through a FXO

while "Long distance" 10-digit dials through a low-cost SIP trunk). Users can also set up a failover trunk to be used when the primary trunk fails.

Go to Web GUI->PBX->Basic/Call Routes->Outbound Routes to add and edit outbound rules.

•

Click on "Create New Outbound Rule" to add a new outbound route.

•

Click on to edit the outbound route.

•

Click on to delete the outbound route.

•

On the UCM6100, the outbound route priority is based on “Best matching pattern”. For example, the

UCM6100 has outbound route A with pattern 1xxx and outbound route B with pattern 10xx configured.

When dialing 1000 for outbound call, outbound route B will always be used first. This is because pattern 10xx is a better match than pattern 1xxx. Only when there are multiple outbound routes with the same pattern configured, users can click on to move the outbound route up/down to

arrange the priority among those outbound routes.

Table 57: Outbound Route Configuration Parameters

Calling Rule Name

Pattern

Password

Call Duration Limit

Configure the name of the calling rule (e.g., local, long_distance, and etc).

Letters, digits, _ and - are allowed.

•

All patterns are prefixed with the "_".

•

Special characters:

X: Any Digit from 0-9.

Z: Any Digit from 1-9.

N: Any Digit from 2-9.

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

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

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

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

Enable to configure the maximum duration for the call using this outbound route.


Maximum Call Duration

Warning Time

Warning Repeat Interval

Privilege Level

Enable Filter on Source Caller

ID

Configure the maximum duration of the call (in seconds). The default setting is 0, which means no limite.

Configure the warning time for the call using this outbound route. If set to x seconds, the warning tone will be played to the caller when x seconds are left to end the call.

Configure the warning repeat interval for the call using this outbound route. If set to x seconds, the warning tone will be played every x seconds after the first warning.

Select privilege level for the outbound rule.

•

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

•

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

•

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

•

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

•

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.

2. Custom Dynamic Route: define the pattern for the source caller ID.

This allows users to define extension range instead of selecting them one by one.

•


•

Special characters:









Send This Call Through Trunk

Use Trunk Select the trunk for this outbound rule.

Strip

Prepend

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

Example:

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

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

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

Use Failover Trunk

Failover Trunk

Strip

Prepend

Failover trunks can be used to make sure that a call goes through an alternate route, when the primary trunk is busy or down. If "Use Failover

Trunk" is enabled and "Failover trunk" is defined, the calls that cannot be placed via the regular trunk may have a secondary trunk to go through.

Example:

The user's primary trunk is a VoIP trunk and the user would like to use the

PSTN when the VoIP trunk is not available. The PSTN trunk can be configured as the failover trunk of the VoIP trunk.

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

Example:

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

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

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

INBOUND ROUTES

Inbound routes can be configured via Web GUI->PBX->Basic/Call Routes->Inbound Routes.

•

Click on "Create New Inbound Rule" to add a new inbound route.

•

Click on "Blacklist" to configure blacklist for all inbound routes.

•

Click on to edit the inbound route.

Page 172 of 304

•

Click on to delete the inbound route.

INBOUND RULE CONFIGURATIONS

Table 58: Inbound Rule Configuration Parameters

Trunks

DID Pattern

Privilege Level

Default Destination

Select the trunk to configure the inbound rule.

•


•

Special characters:







•

The pattern can be composed of two parts, divided by a ‘/’ character.

The first part is used to specify the dialed number the second part is used to specify the caller ID and it is optional, if set it means only the extension with the specific caller ID is allowed to call in or call out.

For example, patter '_2XXX/1234' means the only extension with the caller ID '1234' is allowed to use this rule.

Configure the privilege level for this inbound route.

Select the default destination for the inbound call.

•

Extension

•

Voicemail

•

Conference Room

•

Queue

•

Ring Group

•

Paging/Intercom

•

Voicemail Group

•

Fax

•

DISA

•

IVR

•

Dial By Name

•

External Number

•

By DID

When "By DID" is used, the UCM6100 will look for the destination based on the number dialed, which could be local extensions, conference, call queue, ring group, paging/intercom group, IVR, voicemail groups and Fax extension as configured in "DID


Strip

Prepend

Prepend Trunk Name

Dial Trunk

DID Destination 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.

This option shows up only when “By DID” is selected. If enabled, the trunk name will be prepended to the display name.

This option shows up only when "By DID" is selected. If enabled, the external users dialing in to the trunk via this inbound route can dial outbound call using the UCM6100’s trunk.

This option shows up only when "By DID" is selected. This controls the destination that can be reached by the external caller via the inbound route. The DID destination are:

•

Extension

•

Conference

•

Call Queue

•

Ring Group

•

Paging/Intercom Groups

•

IVR

•

Voicemail Groups

•

Fax Extension

•

Dial By Name

•

All

When present in an INVITE request, the Alert-Info header field specifies an alternative ring tone to the UAS.

Alert-Info

Time Condition

Time Conditions

Destination

Select the time condition for the inbound rule.

Select the destination for the inbound call during the specified time condition.

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 98: Inbound Route feature: Prepend

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 then prepend a 2 to the stripped number, now the number become 2163.

4. UCM6100 will now forward the incoming call to extension 2163.

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 to add to the list.

•

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


Figure 99: Blacklist Configuration Parameters

•

To add blacklist number in batch, click on to upload blacklist file in csv format. The supported csv format is as below.

Figure 100: 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->PBX->Internal Options->Feature Codes.

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CONFERENCE BRIDGE

The UCM6100 supports conference bridge allowing multiple bridges used at the same time:

•

UCM6102/6104 supports up to 3 conference bridges allowing up to 25 simultaneous PSTN or IP participants.

•

UCM6108/6116 supports up to 6 conference bridges allowing up to 32 simultaneous PSTN or IP participants.

The conference bridge configurations can be accessed under Web GUI->PBX->Call

Features->Conference. In this page, users could create, edit, view, invite, manage the participants and delete conference bridges. The conference bridge status and conference call recordings (if recording is

enabled) will be displayed in this web page as well.

CONFERENCE BRIDGE CONFIGURATIONS

•

Click on "Create New Conference Room" to add a new conference bridge.

•

Click on to edit the conference bridge.

•

Click on to delete the conference bridge.

Extension

Password

Admin Password

Table 59: Conference Bridge Configuration Parameters

Configure the conference number for the users to dial into the conference.

When configured, the users who would like to join the conference call must enter this password before accessing the conference bridge.

Note:

•

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

•

The password has to be at least 4 characters.

Configure the password to join the conference bridge as administrator.

Conference administrator can manage the conference call via IVR (if

"Enable Caller Menu" is enabled) as well as invite other parties to join the conference by dialing "0" (permission required from the invited party) or

"1" (permission not required from the invited party) during the conference call.


Enable Caller Menu

Record Conference

Quiet Mode

Wait For Admin

Enable User Invite

Announce Callers

Public Mode

Play Hold Music


Note:

•

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

•


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

If enabled, the calls in this conference bridge will be recorded automatically in a .wav format file. All the recording files will be displayed and can be downloaded in the conference web page. The default setting is "No".

If enabled, if there are users joining or leaving the conference, voice

prompt or notification tone won't be played. The default setting is "No".

Note:

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

If enabled, the participants will not hear each other until the conference

administrator joins the conference. The default setting is "No".

Note:

If "Quiet Mode" is enabled, the voice prompt for "Wait For Admin" will not be announced.

If enabled, users could press 0 to invite other users (with the users' permission) or press 1 to invite other users (without the user's permission) to join the conference. The default setting is "No".

Note:

Conference administrator can always invite other users without enabling this option.

If enabled, the caller will be announced to all conference participants

when there the caller joins the conference. The default setting is "No".

Note:

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

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

If enabled, the UCM6100 will play Hold music when there is only one user


Music On Hold

Skip Authentication When

Inviting User via Trunk from

Web GUI

in the conference. The default setting is "No".

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

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

On Hold.

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

JOIN A CONFERENCE CALL

Users could dial the conference bridge extension to join the conference. If password is required, enter the password to join the conference as a normal user, or enter the admin password to join the conference as administrator.

INVITE OTHER PARTIES TO JOIN CONFERENCE

When using the UCM6100 conference bridge, there are two ways to invite other parties to join the

conference.

•

Invite from Web GUI.

For each conference bridge in UCM6100 Web GUI->PBX->Call Features->Conference, there is an icon

for option "Invite a participant". Click on it and enter the number of the party you would like to invite.

Then click on "Add". A call will be sent to this number to join it into the conference.

Figure 101: Conference Invitation From Web GUI

•

Invite by dialing 0 or 1 during conference call.

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


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

0: If 0 is entered to invite other party, once the invited party picks up the invitation call, a permission will be

asked to "accept" or "reject" the invitation before joining the conference.

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

--------------------------------------------------------------------------------------------------------------------------------------------

Note:

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

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

--------------------------------------------------------------------------------------------------------------------------------------------

DURING THE CONFERENCE

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

•

Manage the conference call from Web GUI.

Log in UCM6100 web GUI during the conference call, the participants in each conference bridge will be listed.

1. Click on to kick a participant from the conference.

2. Click on to mute the participant.

3. Click on to lock this conference bridge so that other users cannot join it anymore.

4. Click on to invite other users into the conference bridge.

•

Manage the conference call from IVR.

If "Enable Caller Menu" is enabled, conference participant can input * to enter the IVR menu for the conference. Please see options listed in the table below.

Table 60: Conference Caller IVR Menu

Conference Administrator IVR Menu


3

4

1

2

5

6

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.

7

8

5

6

1

4

Increase your volume.

More options.

•

1: List all users currently in the conference call.

•

2: Kick all non-Administrator participants from the conference call.

•

3: Mute/Unmute all non-Administrator participants from the conference call.

•

4: Record the conference call.

•

8: Exit the caller menu and return to the conference.

Conference User IVR Menu

Mute/unmute yourself.

Decrease the volume of the conference call.

Decrease your volume.

Increase the volume of the conference call.

7

8

Increase your volume.

Exit the caller menu and return to the conference.

--------------------------------------------------------------------------------------------------------------------------------------------

Note:

When there is participant in the conference, the conference bridge configuration cannot be modified.

--------------------------------------------------------------------------------------------------------------------------------------------

RECORD CONFERENCE

The UCM6100 allows users to record the conference call and retrieve the recording from web

GUI->PBX->Call Features->Conference.

To record the conference call, when the conference bridge is in idle, enable "Record Conference" from the conference bridge configuration dialog. Save the setting and apply the change. When the conference call starts, the call will be automatically recorded in .wav format.


The recording files will be listed as below once available. Users could click on to download the recording or click on to delete the recording. 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 102: Conference Recording


IVR

CONFIGURE IVR

IVR configurations can be accessed under the UCM6100 Web GUI->PBX->Call Features->IVR. Users could create, edit, view and delete an IVR.

•

Click on "Create New IVR" to add a new IVR.

•

Click on to edit the IVR configuration.

•

Click on to delete the IVR.

Table 61: IVR Configuration Parameters

Basic Settings

Name

Extension

Dial Other Extensions

Dial Trunk

Permission

Welcome Prompt

Digit Timeout

Response Timeout

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

Enter the extension number for users to access the IVR.

If enabled, all callers to the IVR can dial other extensions. The default setting is "No".

If enabled, all callers to the IVR is allowed to use trunk. The permission must be configured for the users to use the trunk first. The default setting is "No".

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

"International" from the lowest level to the highest level. The default setting is "Internal". If the user tries to dial outbound calls after dialing into the IVR, the UCM6100 will compared the IVR's permission level with the outbound route's privilege level. If the IVR's permission level is higher than (or equal to) the outbound route's privilege level, the call will be allowed to go through.

Select an audio file to play as the welcome prompt for the IVR. Click on

"Prompt" to add additional audio file under web GUI->Internal

Options->IVR Prompt.

Configure the timeout between digit entries. After the user enters a digit, the user needs to enter the next digit within the timeout. If no digit is detected within the timeout, the UCM6100 will consider the entries complete. The default timeout is 3 seconds.

After playing the prompts in the IVR, the UCM6100 will wait for the DTMF entry within the timeout (in seconds). If no DTMF entry is detected within


Response Timeout Prompt

Invalid Prompt

Response Timeout Repeat

Loops

Invalid Repeat Loops

Language

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 *

Timeout

Invalid

the timeout, a timeout prompt will be played. The default setting is 10 seconds.

Select the prompt message to be played when timeout occurs.

Select the prompt message to be played when an invalid extension is pressed.

Configure the number of times to repeat the prompt if no DTMF input is detected. When the loop ends, it will go to the timeout destination if configured, or hang up. The default setting is 3.

Configure the number of times to repeat the prompt if the DTMF input is invalid. When the loop ends, it will go to the invalid destination if configured, or hang up. The default setting is 3.

Select the voice prompt language to be used for this IVR. The default setting is "Default" which is the selected voice prompt language under web GUI->PBX->Internal Options->Language. The dropdown list shows all the current available voice prompt languages on the UCM6100.

To add more languages in the list, please download voice prompt package by selecting "Check Prompt List" under web

GUI->PBX->Internal Options->Language.

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

•

Extension

•

Voicemail

•

Conference Rooms

•

Voicemail Group

•

IVR

•

Ring Group

•

Queues

•

Page Group

•

Fax

•

Custom Prompt

•

Hangup

•

DISA

•

Dial By Name

•

External Number

•

Callback


CREATE CUSTOM PROMPT

To record new IVR prompt or upload IVR prompt to be used in IVR, click on “Prompt” next to the “Welcome

Prompt” option and the users will be redirected to Custom Prompt page. Or users could go to Web

GUI->PBX->Internal Options->Custom Prompt page directly.

Figure 103: Click on Prompt to Create IVR Prompt

Once the IVR prompt file is successfully added to the UCM6100, it will be added into the prompt list options for users to select in different IVR scenarios.

RECORD NEW CUSTOM PROMPT

Settings

In the UCM6100 web UI->PBX->Internal Options->Custom Prompt page, click on “Record New Custom

Prompt” and follow the steps below to record new IVR prompt.

Figure 104: Record New Custom Prompt

•

Specify the IVR file name.

•

Select the format (GSM or WAV) for the IVR prompt file to be recorded.

•

Select the extension to receive the call from the UCM6100 to record the IVR prompt.


•

Click the “Record” button. A request will be sent to the UCM6100. The UCM6100 will then call the extension for recording the IVR prompt from the phone.

•

Pick up the call from the extension and start the recording following the voice prompt.

•

The recorded file will be listed in the IVR Prompt web page. Users could select to re-record, play or delete the recording.

UPLOAD CUSTOM PROMPT

If the user has a pre-recorded IVR prompt file, click on “Upload Custom Prompt” in Web

GUI->PBX->Internal Options->Custom Prompt page to upload the file to the UCM6100. The following are required for the IVR prompt file to be successfully uploaded and used by the UCM6100:

•

PCM encoded.

•

16 bits.

•

8000Hz mono.

•

In .mp3 or .wav format;; or raw/ulaw/alaw/gsm file with .ulaw or .alaw suffix.

•

File size under 5M.

Figure 105: Upload Custom Prompt

Click on to select audio file from local PC and click on to start uploading. Once uploaded, the file

will appear in the Custom Prompt web page.


LANGUAGE SETTINGS FOR VOICE PROMPT

The UCM6100 supports multiple languages in web GUI as well as system voice prompt. Currently, there are 16 languages supported in system voice prompt: English (United States), Arabic, Chinese, Dutch,

English (United Kingdom), French, German, Greek, Hebrew, Italian, Polish, Portuguese, Russian,

Spanish, Swedish and Turkish.

English (United States) and Chinese voice prompts are built in with the UCM6100 already. The other languages provided by Grandstream can be downloaded and installed from the UCM6100 web GUI directly. Additionally, users could customize their own voice prompts, package them and upload to the

UCM6100.

Language settings for voice prompt can be accessed under Web GUI->PBX->Internal


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 106: Language Settings for Voice Prompt

A new dialog window of voice prompt package list will be displayed. Users can see the version number

(latest version available V.S. current installed version), package size and options to upgrade or download the language.


Figure 107: 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 108: 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.


VOICEMAIL

CONFIGURE VOICEMAIL

If the voicemail is enabled for UCM6100 extensions, the configurations of the voicemail can be globally set up and managed under Web GUI->PBX->Call Features->Voicemail.

Table 62: Voicemail Settings

Max Greeting

Dial ‘0’ For Operator

Max Messages Per Folder

Max Message Time

Min Effective Message Time

Announce Message Caller-ID

Configure the maximum number of seconds for the voicemail greeting.

The default setting is 60 seconds.

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

Configure the maximum number of messages per folder in users’ voicemail. The valid range 10 to 1000. The default setting is 50.

Select the maximum duration of the voicemail message. The message will not be recorded if the duration exceeds the max message time. The default setting is 15 minutes. The available options are:

•

1 minute

•

2 minutes

•

5 minutes

•

15 minutes

•

30 minutes

•

Unlimited

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

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

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

•

No minimum

•

1 second

•

2 seconds

•

3 seconds

•

4 seconds

•

5 seconds

Note:

Silence and noise duration are not counted in message time.

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


Announce Message Duration

Play Envelope

Play from Last 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".

Allow User Review

ACCESS VOICEMAIL

If the voicemail is enabled for UCM6100 extensions, the users can dial the voicemail access feature code

(by default *98 or *97) to access the extension’s voicemail. The users will be prompt to enter the voicemail password and then can enter digits from the phone keypad to navigate in the IVR menu for different options.

Table 63: Voicemail IVR Menu

Main Menu Sub Menu 1

1 - New messages

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

2 - Change folders

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


3 -

Advanced options

# - Cancel

1 - Send a reply

2 - Call the person who sent this message

3 - Hear the message envelop

4 - Leave a message


1 - Record your unavailable message

2 - Record your busy message

0 - Mailbox options

3 - Record your name

4 - Record temporary greeting

1 - Accept this recording

2 - Listen to it

3 - Re-record your message


2 - Listen to it



2 - Listen to it



2 - Listen to it


5 - Change your password


VOICEMAIL EMAIL SETTINGS

The UCM6100 can be configured to send the voicemail as attachment to Email. Click on "Voicemail Email

Settings" button to configure the Email attributes and content.

Table 64: Voicemail Email Settings

Attach Recordings to E-Mail

Keep Recordings

Template For Voicemail Emails

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

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 109: Voicemail Email Settings

Click on "Load Default Settings" button to view the default template as an example.

CONFIGURE VOICEMAIL GROUP

The UCM6100 supports voicemail group and all the extensions added in the group will receive the voicemail to the group extension. The voicemail group can be configured under Web GUI->PBX->Call

Features->Voicemail Group. Click on "Create New Voicemail Group" to configure the group.


Extension

Name

Voicemail Password

Email Address

Voicemail Group Mailboxes

Figure 110: Voicemail Group

Table 65: 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.



RING GROUP

The UCM6100 supports ring group feature with different ring strategies applied to the ring group members.

This section describes the ring group configuration on the UCM6100.

CONFIGURE RING GROUP

Ring group settings can be accessed via Web GUI->PBX->Call Features->Ring Group.

Figure 111: Ring Group

•

Click on “Create New Ring Group” to add ring group.

•

Click on to edit the ring group. The following table shows the ring group configuration parameters.

•

Click on to delete the ring group.

Ring Group Name

Extension

Ring Group Members

Selected LDAP Numbers

Ring Strategy

Table 66: 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.


Permission

Custom Prompt

Ring Timeout on Each Member

Auto Record

Enable Destination

Secret

Email Address

Configure the permission level for Ring Group. This permission will take effect when an incoming call is going to be directed to an external number configured as “Default Destination”. The available permissions are

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

This option is to set a custom prompt for a ring group to announce to caller. Click on ‘Prompt’, it will direct to the page PBX->Internal

Options->Custom Prompt, where users could record new prompt or upload prompt files.

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


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

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

Note:


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

"Attach Recordings to E-mail" is enabled from Web

GUI->PBX->Voicemail->Voicemail Email Settings, the voicemail can be sent to the ring group's Email address as attachment.


Figure 112: 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 UI-> PBX-> Basic/Call Routes-> VoIP 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.


Figure 113: 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.

Figure 114: Manually Sync LDAP Server

4. Under Ring Groups setting page, click under web UI-> PBX-> Call Features-> Ring Groups.

. Ring Groups can be found


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 115: Ring Group Remote Extension



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 sections describes the

configuration of paging/intercom group under Web GUI->PBX->Call Features->Paging/Intercom.

CONFIGURE PAGING/INTERCOM GROUP

•

Click on "Create New Paging/Intercom Group" to add paging/intercom group.

Name

Extension

Type

Custom Prompt

Page/Intercom Group

Members

Figure 116: Paging/Intercom Group

Table 67: Paging/Intercom Group Configuration Parameters

Configure paging/intercom group name.

Configure the paging/intercom group extension.

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

This option is to set a custom prompt for a paging/intercom group to announce to caller. Click on ‘Prompt’, it will direct to the page

PBX->Internal Options->Custom Prompt, where users could record new prompt or upload prompt files.

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


•

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 117: Page/Intercom Group Settings

•

The UCM6100 has pre-configured paging/intercom feature code. By default, the Paging Prefix is *81 and the Intercom Prefix is *80. To edit page/intercom feature code, click on "Feature Codes" in the

"Paging/Intercom Group Settings" dialog. Or users could go to Web GUI->PBX->Internal

Options->Feature Codes directly.


CALL QUEUE

The UCM6100 supports call queue by using static agents or dynamic agents. 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->PBX->Call

Features->Call Queue.

CONFIGURE CALL QUEUE

Call queue settings can be accessed via Web GUI->PBX->Call Features->Call Queue.

Figure 118: Call Queue

•

Click on "Create New Queue" to add call queue.

•

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

Extension

Name

Strategy

Table 68: Call Queue Configuration Parameters

Configure the call queue extension.

Configure the call queue name to identify the call queue.

Select the strategy for the call queue.

•

Ring All

Ring all available Agents simultaneously until one answers.

•

Linear

Ring agents in the specified order.


Music On Hold

Leave When Empty

Dial in Empty Queue

Permission


•

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->Internal

Options->Music On Hold.

Configure whether the callers will be disconnected from the queue or not if the queue has no agent anymore. The default setting is "Strict".

•

Yes

Callers will be disconnected from the queue if all agents are paused or invalid.

•

No

Never disconnect the callers from the queue when the queue is empty.

•

Strict

Callers will be disconnected from the queue if all agents are paused, invalid or unavailable.

Configure whether the callers can dial into a call queue if the queue has no agent. The default setting is "No".

•

Yes

Callers can always dial into a call queue.

•

No

Callers cannot dial into a queue if all agents are paused or invalid.

•

Strict

Callers cannot dial into a queue if the agents are paused, invalid or unavailable.

Configure the permission level for Call Queue. This permission will take effect when an incoming call is going to be directed to an external number configured as “Default Destination”. The available permissions are

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


Dynamic Login Password

Ring Time Out

Wrapup Time

Max Queue Length

Report Hold Time

Wait Time

Auto Record

Enable Destination

Queue Timeout

Failover Destination

Enable Feature Codes

Agents


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

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

Configure the number of seconds before a new call can ring the queue after the last call on the agent is completed. If set to 0, there will be no delay between calls to the queue. The default setting is 15 seconds.

Configure the maximum number of calls to be queued at once. This number does not include calls that have been connected with agents. It only includes calls not connected yet. The default setting is 0, which means unlimited. When the maximum value is reached, the caller will be treated with busy tone followed by the next calling rule after attempting to enter the queue.

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

"No".

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

Note:

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

Time".

If enabled, the calls on the call queue will be automatically recorded. The recording files can be accessed in Queue Recordings under web

GUI->PBX->Call Features->Call Queue.

If enabled, the incoming call for the call queue will be routed to the destination configured in the next field if none of the agents answers the call after ringing for a time of “Ring Timeout”.

Configure the global timeout (in seconds) of call queue. It must be bigger than the value of ring timeout. The call in the queue will be transferred to the failover destination directly if this time is exceeded.

Configure the call destination for the call to be routed to if no agent in this call queue answers the call.

Enable feature codes option for call queue. For example, *83 is used for

“Agent Pause”

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

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


•

Click on to delete the call queue.

•

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

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

Figure 119: Agent Login Settings

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

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

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

•

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

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

•

Queue recordings are shown on the Call Queue page. Click on to download the recording file in

.wav format;; click on to delete the recording file. To delete multiple recording files by one click, select several recording files to be deleted and click on “Delete Selected Recording Files” or click on

“Delete All Recording Files” to delete all recording files.


EXTENSION GROUPS

The UCM6100 extension group feature allows users to assign and categorize extensions in different groups to better manage the configurations on the UCM6100. For example, when configuring "Enable

Filter on Source Caller ID", users could select a group instead of each person's extension to assign. This

feature simplifies the configuration process and helps manage and categorize the extensions for business environment.

CONFIGURE EXTENSION GROUPS

Extension group can be configured via Web GUI->PBX->Call Features->Extension Groups.

•

Click on "Create New Extension Group" to create a new extension group.

•

Click on to edit the extension group.

Select extensions from the list on the left side to the right side.

Figure 120: Edit Extension Group

•

Click on to delete the extension group.


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USING EXTENSION GROUPS

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

Routes->Outbound Routes and select "Enable Filter on Source Caller ID". Both single extensions and

extension groups will show up for users to select.

Figure 121: Select Extension Group in Outbound Route


PICKUP GROUPS

The UCM6100 supports pickup group feature which allows users to pick up incoming calls for other

extensions if they are in the same pickup group, by dialing "Pickup Extension" feature code (by default *8).

CONFIGURE PICKUP GROUPS

Pickup groups can be configured via Web GUI->PBX->Call Features->Pickup Groups.

•

Click on "Create New Pickup Group" to create a new pickup group.

•

Click on to edit the pickup group.

Select extensions from the list on the left side to the right side.

Figure 122: Edit Pickup Group

•

Click on to delete the pickup group.

CONFIGURE PICKUP FEATURE CODE

When picking up the call for the pickup group member, the user only needs to dial the pickup feature code.

It’s not necessary to add the extension number after the pickup feature code. The pickup feature code is configurable under Web GUI->PBX->Internal Options->Feature Codes.

The default pickup feature code is *8.


Figure 123: Edit Pickup Feature Code


MUSIC ON HOLD

Music On Hold settings can be accessed via Web GUI->PBX->Internal Options->Music On Hold. In this page, users could configure music on hold class and upload music files. The "default" Music On Hold class already has 5 audio files defined for users to use.

Figure 124: Music On Hold Default Class

•

Click on "Create New MOH Class" to add a new Music On Hold class.

•

Click on to configure the MOH class sort method to be "Alpha" or "Random" for the sound files.

•

Click on next to the selected Music On Hold class to delete this Music On Hold class.


•

Click on to select music file from local PC and click on to start uploading. The music file uploaded has to be 8 KHz Mono format with size smaller than 5M.

•

Click on next to the sound file to delete it from the selected Music On Hold Class.

•

Select the sound files and click on to delete all selected music on hold files.

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Note:

Once the MOH file is deleted, there are two ways to recover the music files.

•

Users could download the MOH file from this link: http://downloads.asterisk.org/pub/telephony/sounds/releases/asterisk-moh-opsound-wav-2.03.tar.gz

After downloading and unzip the pack, users could then upload the music files to UCM.

•

Factory reset could also recover the MOH file on the UCM.

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FAX/T.38

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

GUI->PBX->Internal Options->FAX/T.38. The list of received Fax files will be displayed in the same web

page for users to view, retrieve and delete.

CONFIGURE FAX/T.38

•

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

•

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

Table 69: FAX/T.38 Settings

Enable Error Correction Mode

Maximum Transfer Rate

Minimum Transfer Rate

Default Email Address

Template Variables

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

Configure the maximum transfer rate during the Fax rate negotiation. The possible values are 2400, 4800, 7200, 9600, 12000 and 14400. The default setting is 14400.

Configure the minimum transfer rate during the Fax rate negotiation. The possible values are 2400, 4800, 7200, 9600, 12000 and 14000. The default setting is 2400.

Configure the Email address to send the received Fax to if user's Email address cannot be found.

Note:

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

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

The template variables are:

•

${CALLERIDNUM} : Caller ID Number

•

${CALLERIDNAME} : Caller ID Name

•

${RECEIVEEXTEN} : The extension to receive the Fax

•

${FAXPAGES} : Number of pages in the Fax

•

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


•

Click on to edit the Fax extension.

•

Click on to delete the Fax extension.

SAMPLE CONFIGURATION TO RECEIVE FAX FROM PSTN LINE

The following instructions 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->PBX->Analog Trunks page.

4. Create or edit the analog trunk for Fax as below.

Fax Mode: Make sure "Fax Mode" option is set to "None".

Figure 125: Configure Analog Trunk without Fax Detection

5. Go to UCM6100 web GUI->PBX->Basic/Call Routes->Extensions page.

6. Create or edit the extension for FXS port.


•

Analog Station: Select FXS port to be assigned to the extension. By default, it's set to "None".

•

Once selected, analog related settings for this extension will show up in "Analog Settings" section.

Figure 126: Configure Extension for Fax Machine

7. Go to web GUI->PBX->Basic/Call Routes->Inbound Routes page.

8. Create an inbound route to use the Fax analog trunk. Select the created extension for Fax machine in step 4 as the default destination.


Figure 127: Configure Inbound Rule for Fax


Page 216 of 304

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->Internal Options->Fax/T.38 page. Create a new Fax extension.

Figure 128: Create Fax Extension

3. Go to UCM6100 web GUI->Basic/Call Routes->Analog Trunks page. Create a new analog trunk.

Please make sure "Fax Detection" is set to "No".

4. Go to UCM6100 web GUI->Basic/Call Routes->Inbound Routes page. Create a new inbound route and set the default destination to the Fax extension.

Figure 129: 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.


ASTERISK MANAGER INTERFACE (RESTRICTED ACCESS)

Starting from firmware 1.0.8.11, the UCM6100 supports Asterisk Manager Interface (AMI) with restricted access. AMI allows a client program to connect to an Asterisk instance commands or read events over a

TCP/IP stream. It’s particularly useful when the system admin tries to track the state of a telephony client

inside Asterisk.

User could configure AMI parameters on UCM6100 web GUI->PBX->Internal Options->AMI. For details

on how to use AMI on UCM6100, please refer to the following AMI guide:

http://www.grandstream.com/products/ucm_series/ucm61xx/documents/ucm6100_ami_guide.pdf

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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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BUSY CAMP-‐‑ON

pdf

Starting from firmware 1.0.8.11, the UCM6100 supports busy camp-on/call completion feature that allows the PBX to camp on a called party and inform the caller as soon as the called party becomes available given the previous attempted call has failed.

The configuration and instructions on how to use busy camp-on/call completion feature can be found in the following guide:

http://www.grandstream.com/products/ucm_series/ucm61xx/documents/ucm6100_busy_camp_on_guide.



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 web GUI-> PBX-> Call Features->Follow Me.

To configure follow me:

•

Click on "Create New Follow Me" and then select an extension to be configured with Follow Me.

Figure 130: Create Follow Me

•

Click on “Next” to continue editing Follow Me configuration.


Figure 131: Edit Follow Me


Page 222 of 304

•

Click on “Add Follow Me Number” to add local extensions or external numbers to be called after ringing the extension selected in the first step.

•

Once created, it will be displayed on the follow me web page list. Click on to edit the Follow Me

configuration. Click on to delete the Follow Me.

The following table shows the Follow Me configuration parameters.

Table 70: Follow Me Settings

Enable

Skip Trunk Auth

Music On Hold Class

Confirm When Answering

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.

Follow Me Numbers

New Follow Me Number

The added numbers are listed here. Click on to arrange the order. Click on to delete the number. Click on to add new numbers.

Add a new Follow Me number which could be a ‘Local Extension’ or

‘External Number’. The selected dial plan should have permissions to dial the defined external number.

Dialing Order

Select the order in which the Follow Me destinations will be dialed to reach the user: ring all at once or ring one after the other.

•

Click on “Follow Me Options” to enable or disable the options listed in the following table.

Table 71: 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.


ONE-‐‑KEY DIAL

The UCM6100 supports One-Key Dial that allows users to call a certain destination by pressing one digit 0

to 9 on the keypad. This creates a system-wide speed dial access for all the extensions on the UCM6100.

To enable One-Key Dial, on the UCM6100 web GUI, go to page PBX->Call Features->One-Key Dial.

Figure 132: Configure One-Key Dial

User should first decide a digit used for One-Key Dial and check the option “Enable Destination” for the digit. Then select a dial destination from “Default Destination”. The supported destinations include extension, voicemail, conference room, voicemail group, IVR, ring group, call queue, page group, fax,

DISA, Dial by Name and external number.


Figure 133: One-Key Dial Destinations


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 Features->DISA.

•

Click on "Create New IVR" to add a new DISA.

•

Click on to edit the DISA configuration.

•

Click on to delete the DISA.

Name

Password


Figure 134: Create New DISA

Table 72: 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.


Permission

Response Timeout

Digit Timeout

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.

Allow Hangup

If enabled, during an active call, users can enter the UCM6100 hangup feature code (by default it's *0) to disconnect the call or hang up directly. A new dial tone will be heard shortly for the user to make a new call. The default setting is "No".

Once successfully created, users can configure the inbound route destination as "DISA" or IVR key event as "DISA". When dialing into DISA, users will be prompted with password first. After entering the correct

password, a second dial tone will be heard for the users to dial out.


CALLBACK FEATURE

Callback is mainly designed for users who often use their mobile phones to make long distance or international calls which may have high service charges. The callback feature provides an economic solution for reduce the cost from this.

The callback feature works as follows:

1. Configure a new callback on the UCM6100.

2. On the UCM6100, configure destination of the inbound route for analog trunk to callback.

3. Save and apply the settings.

4. The user calls the PSTN number of the UCM6100 using the mobile phone, which goes to callback destination as specified in the inbound route.

5. Once the user hears the ringback tone from the mobile phone, hang up the call on the mobile phone.

6. The UCM6100 will call back the user.

7. The user answers the call.

8. The call will be sent to DISA or IVR which directs the user to dial the destination number.

9. The user will be connected to the destination number.

In this way, the calls are placed and connected through trunks on the UCM6100 instead of to the mobile

phone directly. Therefore, the user will not be charged on mobile phone services for long distance or international calls.

To configure callback on the UCM6100, go to web GUI->PBX->Call Features->Callback page and click

on . Configuration parameters are listed in the following table.

Name

CallerID Pattern

Outbound Prepend

Table 73: Callback Configuration Parameters

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.


Page 228 of 304

Delay Before Callback

Destination

Configure the number of seconds to be delayed 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 UCM6100 trunk.


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 setting

is under web GUI->Call Features->Event List.

•

Click on "Create New Event List" to add a new event list.

•

Click on to edit the event list configuration.

•

Click on to delete the event list.

URI

Local Extensions

Remote Extensions

Table 74: 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,


Special Extensions

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

Figure 135: Create New Event List

Remote extension monitoring works on the UCM6100 via event list BLF, among Peer SIP trunks or

Register SIP trunks (register to each other). Therefore, please properly configure SIP trunks on the

UCM6100 first before using remote BLF feature. Please note the SIP end points need support event list

BLF in order to monitor remote extensions.

When an event list is created on the UCM6100 and remote extensions are added to the list, the UCM6100 will send out SIP SUBSCIRBE to the remote UCM6100 to obtain the remote extension status. When the

SIP end points registers and subscribes to the local UCM6100 event list, it can obtain the remote extension status from this event list. Once successfully configured, the event list page will show the status of total extension and subscribers for each event list. Users can also select the event URI to check the

monitored extension's status and the subscribers' details.

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Note:

•

To configure LDAP sync, please go to UCM6100 web GUI->PBX->Basic/Call Routes->VoIP Trunk.

You will see "Sync LDAP Enable" option. Once enabled, please configure password information for the remote peer UCM6100 to connect to the local UCM6100. Additional information such as port number,

LDAP outbound rule, LDAP Dialed Prefix will also be required. Both the local UCM6100 and remote

UCM6100 need enable LDAP sync option with the same password for successful connection and synchronization.

•

Currently LDAP sync feature only works between two UCM6100s.

•

(Theoretically) Remote BLF monitoring will work when the remote PBX being monitored is non-UCM6100 PBX. However, it might not work the other way around depending on whether the non-UCM6100 PBX supports event list BLF or remote monitoring feature.

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DIAL BY NAME

Dial By Name is a feature on the PBX that allows caller to search a person by first or last name via his/her phone's keypad. The administrator can define the Dial By Name directory including the desired extensions in the directory and the searching type by "first name" or "last name". After dialing in, the PBX IVR/Auto

Attendant will guide the caller to spell the digits to find the person in the Dial By Name directory. This feature allows customers/clients to use the guided automatic system to get in touch with the enterprise employees without having to know the extension number, which brings convenience and improves

business image for the enterprise.

DIAL BY NAME CONFIGURATION

The administrators can create the dial by name group under web GUI->PBX->Call Features->Dial By

Name.


Figure 136: Create Dial By Name Group


Page 234 of 304

1. Group Name

Enter the Group Name. This is to identify the Dial By Name group. The Dial By Name group can be used as the destination for inbound route and key pressing event for IVR. The group name defined here will show up in the destination list when configuring IVR and inbound route. 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 137: Dial By Name Group In IVR Key Pressing Events


Figure 138: Dial By Name Group In Inbound Rule

2. Extension

Configure the direct dial extension for the Dial By Name group.

3. Available Extensions/Selected Extensions

Select available extensions from the left side to the right side as the directory for the Dial By Name group. Only the selected extensions here can be reached by the Dial By Name IVR when dialing into this group. The extensions here must have a valid first name and last name configured under web

GUI->PBX->Basic/Call Routes->Extensions in order to be searchable in Dial By Name directory through IVR. By specifying the extensions here, the administrators can make sure unscreened calls will not reach the company employee if he/she doesn't want to receive them directly.


Figure 139: Configure Extension First Name and Last Name

4. Query Type

Specify the query type. This defines how the caller will need to enter to search the directory.

By First Name: enter the first 3 digits of the first name to search the directory.

By Last Name: enter the first 3 digits of the last name to search the directory.

By Full Name: enter the first 3 digits of the first name or last name to search the directory.

5. Select Type

Specify the select type on the searching result. The IVR will confirm the name/number for the party the caller would like to reach before dialing out.

By Order: After the caller enters the digits, the IVR will announce the first matching party's name and number. The caller can confirm and dial out if it's the destination party, or press * to listen to the next matching result if it's not the desired party to call.

By Menu: After the caller enters the digits, the IVR will announce 8 matching results. The caller can press number 1 to 8 to select and call, or press 9 for results in next page.


ACTIVE CALLS AND MONITOR

The active calls on the UCM6100 are displayed in web UI->Status->Active Calls page. Users can monitor

the status, hang up the call as well as barge in the active calls in real time manner.

ACTIVE CALLS STATUS

To view the status of active calls, navigate to web GUI->Status->Active Calls. The following figure shows

extension 1000 is calling 1001. 1001 is ringing.

Figure 140: Status->PBX Status->Active Calls - Ringing

The following figure shows the call between 1000 and 1001 is established.


Figure 141: Status->PBX Status->Active Calls – Call Established


In active call web page, click on to refresh the active call status.

HANG UP ACTIVE CALLS

icon in the active call dialog. Users can also click on To hang up an active call, click on

to hang up all active calls.

CALL MONITOR

During an active call, click on icon and the monitor dialog will pop up.

Figure 142: Configure to Monitor an Active Call

In the “Monitor” dialog, configure the following to monitor an active call:

1. Enter an available extension for “Monitor’s Extension” which will be used to monitor the active call.

2. “Monitored Extension” must be one of the parties in the active call to be monitored.

3. Select spy mode. There are three options in “Spy Mode”.

•

Listen

In “Listen” mode, the extension monitoring the call can hear both parties in the active call but the audio of the user on this extension will not be heard by either party in the monitored active call.

•

Whisper

In “Whisper” mode, the extension monitoring the call can hear both parties in the active call. The user on this extension can only talk to the selected monitored extension and he/she will not be heard by the other party in the active call. This can be usually used to supervise calls.

•

Barge

In “Barge” mode, the extension monitoring the call can talk to both parties in the active call. The call will be established similar to three-way conference.


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 75: UCM6100 Feature Codes]

and

[ENABLE SPY]

section for instructions.



CALL FEATURES

The UCM6100 supports call recording, transfer, call forward, call park and other call features via feature

code. This section lists all the feature codes in the UCM6100 and describes how to use the call features.

FEATURE CODES

Table 75: UCM6100 Feature Codes

Feature Maps

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

Disconnect

•

Default code: *2.

•

Enter the code during active call. After hearing "Transfer", you will hear the dial tone. Enter the number to transfer to and the user will be connected to this number. Hang up the call to complete the attended transfer.

•

Options

Disable




•

Default code: *0.

•

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

•

Options

Disable





Call Park

Audio Mix Record

•

Default code: #72.

•

Enter the code during active call to park the call.

•

Options

Disable




•

Default code: *3.

•

Enter the code followed by # or SEND to start recording the audio call and the UCM6100 will mix the streams natively on the fly as the call is in progress.

•

Options

Disable




DND/Call Forward

Do Not Disturb (DND) Activate

Do Not Disturb (DND) Deactivate

Call Forward Busy Activate

Call Forward Busy Deactivate

Call Forward No Answer Activate

•

Default code: *77.

•

Default code: *78.

•

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.

•

Default Code: *92.

•



Deactivate


Activate

•

Default Code: *93.

•

Default Code: *72.

•



Deactivate

Feature Misc

•

Default Code: *73.


Feature Code Digits Timeout

Call Park

Parked Lots

Use Parklot as Extension

Parking Timeout (s)

Feature Codes

My Voicemail

Agent Pause

Agent Unpause

Paging Prefix

Voicemail Access Code

Intercom Prefix


•

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.

•

Default Code: *98.

•

Enter *98 and follow the voice prompt. Or dial *98 followed by the extension and # to access the entered extension's voicemail box.

•

Default Code: *97.

•

Press *97 to access the voicemail box.

•

Default Code: *83.

•

Pause the agent in all call queues.

•

Default Code: *84.

•

Unpause the agent in all call queues.

•

Default Code: *81.

•

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

•

Default Code: *80.

•

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


Blacklist Add

Blacklist Remove

Call Pickup on Ringing

Pickup Extension

Direct Dial Voicemail Prefix

Call Completion Request

Call Completion Cancel

Enable Spy

Listen Spy

Whisper Spy


•

Default Code: *40.

•

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

•

Default Code: *41.

•

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

•

Default Code: **.

•

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

•

Default Code: *8.

•

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

•

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

•

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.

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. The default setting is *54.

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.

The default setting is *55.


Barge Spy

This is the feature code to join in on the call to assist both parties. The default setting is *56.

CALL RECORDING

The UCM6100 allows users to record audio during the call. If "Auto Record" is turned on for an extension,

ring group, call queue or trunk, the call will be automatically recorded when there is established call with it.

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

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

2. After establishing the call, enter the "Audio Mix Record" feature code (by default it's *3) followed by # or SEND to start recording.

3. To stop the recording, enter the "Audio Mix Record" feature code (by default it's *3) followed by # or

SEND again. Or the recording will be stopped once the call hangs up.

4. The recording file can be retrieved under Web GUI->Status->CDR. Click on to play the recording or click on to download the recording file.

Figure 143: Download Recording File from CDR Page

The above recorded call's recording files are also listed under the UCM6100 web GUI->CDR->Recording

Files.


Figure 144: 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 THE PARKED CALL

To retrieve the parked call, simply dial the parking lot number and the call will be established. If a parked call is not retrieved after the timeout, the original extension who parks the call will be called back.

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.

--------------------------------------------------------------------------------------------------------------------------------------------

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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INTERNAL OPTIONS

This section describes internal options that haven't been mentioned in previous sections yet. The settings in this section can be applied globally to the UCM6100, including general configurations, jitter buffer, RTP settings, ports config and STUN monitor. The options can be accessed via Web GUI->PBX->Internal

Options.

INTERNAL OPTIONS/GENERAL

Table 76: Internal Options/General

General Preferences

Global OutBound CID

Global OutBound CID Name

Operator Extension

Ring Timeout

Record Prompt

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

CallerID is defined with higher priority. If no CallerID is defined for extension or trunk, the global outbound CID will be used as CallerID.

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

Specify the operator extension, which will be dialed when users presses 0 to exit voicemail application. The operator extension can also be used in

IVR option.

Configure the number of seconds to ring an extension before the call goes to the user's voicemail box. The default setting is 60.

Note:

This is the global value used for each extension if "Ring Timeout" field is left empty on the extension configuration page.

If enabled, users will hear voice prompt before recording is started or stopped. For example, before recording, the UCM6100 will play voice prompt "The call will be recorded". The default setting is "No".

Extension Preferences

Enforce Strong Passwords

If enabled, strong password will be enforced for the password created on

the UCM6100. The default setting is enabled.

Strong Password Rules:

1. Password for voicemail, voicemail group, outbound route, DISA, call queue and conference requires non-repetitive and non-sequential digits, with a minimum length of 4 digits. Repetitive digits pattern


Enable Random Password

Disable Extension Range

Extension Ranges

(such as 0000, 1111, 1234, 2345, and etc), or common digits pattern

(such as 111222, 321321 and etc) are not allowed to be configured as password.

2. Password for extension registration, web GUI admin login, LDAP and

LDAP sync requires alphanumeric characters containing at least two categories of the following, with a minimum length of 4 characters.

•

Numeric digits

•

Lowercase alphabet characters

•

Uppercase alphabet characters

•

Special characters

If enabled, random password will be generated when the extension is created. The default setting is "Yes". It is recommended to enable it for security purpose.

If set to "Yes", users could disable the extension range pre-configured/configured on the UCM6100. The default setting is "No".

Note:

It is recommended to keep the system assignment to avoid inappropriate usage and unnecessary issues.

The default extension range assignment is:

•

User Extensions: 1000-6299

User Extensions is referring to the extensions created under web

UI->PBX->Basic/Call Routes->Extensions page.

•

Pick Extensions: 4000-4999

This refers to the extensions that can be manually picked from end device when being provisioned by the UCM6100. There are two related options in zero config page->Auto Provision Settings, "Pick

Extension Segment" and "Enable Pick Extension". If "Enable Pick

Extension" under zero config settings is selected, the extension list defined in "Pick Extension Segment" will be sent out to the device after receiving the device's request. This "Pick Extension Segment" should be a subset of the "Pick Extensions" range here. This feature is for the GXP series phones that support selecting extension to be provisioned via phone's LCD.

•

Auto Provision Extensions: 5000-6299

This sets the range for "Zero Config Extension Segment" which is the


extensions can be assigned on the UCM6100 to provision the end device.

•

Conference Extensions: 6300-6399

•

Ring Group Extensions: 6400-6499

•

Queue Extensions: 6500-6599

•

Voicemail Group Extensions: 6600-6699

•

IVR Extensions: 7000-7100

•

Fax Extensions: 7200-8200

INTERNAL OPTIONS/JITTER BUFFER

Table 77: Internal Options/Jitter Buffer

SIP Jitter Buffer

Enable Jitter Buffer

Jitter Buffer Size

Max Jitter Buffer

Implementation

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

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

"Fixed" jitter buffer, or used as the initial time for "adaptive" jitter buffer.

The default setting is 100.

Configure the maximum time (in ms) to buffer for "Adaptive" jitter buffer implementation, or used as the jitter buffer size for "Fixed" jitter buffer implementation. The default setting is 200.

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

•

Fixed

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

•

Adaptive

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

INTERNAL OPTIONS/RTP SETTINGS

Table 78: Internal Options/RTP Settings

RTP Start

RTP End

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

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


Strict RTP

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".

RTP Checksums

INTERNAL OPTIONS/PORTS CONFIG

The analog hardware (FXS port and FXO port) on the UCM6100 will be listed in this page. Click on to

edit signaling preference for FXS port or configure ACIM settings for FXO port.

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

Figure 145: FXS Ports Signaling Preference

For FXO port, users could manually enter the ACIM settings by selecting the value from dropdown list for each port. Or users could click on "Detect" for the UCM6100 to automatically detect the ACIM value. The detecting value will be automatically filled into the settings.

Tone Region


FXO Opermode


Figure 146: FXO Ports ACIM Settings

Table 79: Internal Options/Ports Config

Select country to set the default tones for dial tone, busy tone, ring tone and etc to be sent from the FXS port. The default setting is "United States of America (USA)".

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


FXS Opermode

FXS TISS Override

PCMA Override

Boost Ringer

Fast Ringer

Low Power

Ring Detect

FXS MWI Mode

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.

INTERNAL OPTIONS/STUN MONITOR

Table 80: Internal Options/STUN Monitor


STUN Server

Configures the IP address or URL of the STUN server to query. If not specified, STUN is disabled. The default setting is stun.ipvideotalk.com.

Valid format:

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

The default port number is 3478 if not specified.

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

STUN Refresh

INTERNAL OPTIONS/PAYLOAD

The UCM6100 payload type for audio codecs and video codes can be configured here.

Table 81: Internal Options/Payload

AAL2-G.726

DTMF

G.721 Compatible

G.726 iLBC

H.264

H.263P

Configure payload type for ADPCM (G.726, 32kbps, AAL2 codeword packing). The default setting is 112.

Configured payload type for DTMF. The default setting is 101.

Configure to enable/disable G.721 compatible. The default setting is Yes.

Configure the payload type for G.726 if "G.721 Compatible" is disabled.

The default setting is 111.

Configure the payload type for iLBC. The default setting is 97.

Configure the payload type for H.264. The default setting is 99.

Configure the payload type for H.263+. The default setting is 100 103.


IAX SETTINGS

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

IAX SETTINGS/GENERAL

Bind Port

Bind Address

IAX1 Compatibility

No Checksums

Delay Reject

ADSI

Music On Hold Interpret

Table 82: IAX Settings/General

Configure the port number that the IAX2 will be allowed to listen to. The default setting is 4569.

Configure the address that the IAX2 will be forced to bind to. The default setting is 0.0.0.0, which means all addresses.

Select to configure IAX1 compatibility. The default setting is "No".

If selected, UDP checksums will be disabled and no checksums will be calculated/checked on systems supporting this features. The default setting is "No".

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

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

Specify which Music On Hold class this channel would like to listen to when being put on hold. This music class is only effective if this channel has no music class configured and the bridged channel putting the call on hold has no "Music On Hold Suggest" setting.

Specify which Music On Hold class to suggest to the bridged channel when putting the call on hold.

Configure the bandwidth for IAX settings. The default setting is "Low".

Music On Hold Suggest

Bandwidth

IAX SETTINGS/REGISTRATION

Table 83: 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.


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/STATIC DEFENSE

Table 84: IAX Settings/Static Defense

Call Token Optional

Enter a single IP address (e.g., 11.11.11.11) or a range of IP addresses

(11.11.11.11/22.22.22.22) for which call token validation is not required.

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

Max Unvalidated Call Numbers Configure the maximum number of unvalidated calls for all IP addresses.

Call Number Limits Configure to limit the number of calls for a give IP address of IP range.

IP or IP Range

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.


SIP SETTINGS

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

SIP SETTINGS/GENERAL

Realm For Digest

Authentication

Bind UDP Port

Bind IP Address

Allow Guest Calls

Overlap Dialing

Allow Transfer

Enable DNS SRV Lookups on

Outbound Calls

MWI From

Table 85: SIP Settings/General

Configure the host name or domain name for the UCM6100. Realms

MUST be globally unique according to RFC3261. The default setting is

Grandstream.

Configure the UDP port used for SIP. The default setting is 5060.

Configure the IP address to bind to. The default setting is 0.0.0.0, which means binding to all addresses.

If enabled, the UCM6100 allows unauthorized INVITE coming into the

PBX and the call can be made. The default setting is "No".

Warning:

Please be aware of the potential security risk when enabling "Allow Guest

Calls" as this will allow any user with the UCM6100 address to dial into the UCM6100.

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

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

Select to enables DNS SRV lookups on outbound calls from the

UCM6100. The default setting is "Yes".

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

"From:" header as the "name" field. If no "From User" is configured, the

"user" field of the URI in the "From:" header will be filled with this value.

SIP Domain Support

Domain

From Domain

Configure the domain for the UCM6100. Incoming INVITE and REFER messages can be matched against a list of "allowed" domains, each of which can direct the call to a specific context if desired. By default, all domains are accepted and sent to the default context or the context associated with the user/peer placing the call. Register to non-local domains will be automatically denied if a domain list is configured. Up to

10 domains can be added.

Configure the domain in the "From:" header of the SIP message. It may



Auto Domain be required by some providers for authentication.

If enabled, the UCM6100 will add local host name and local IP to domain list. The default setting is "No".

If enabled, requests for external domains that are not served by the

UCM6100 will be allowed. The default setting is "Yes".

Allow External Domains

SIP SETTINGS/MISC

Table 86: SIP Settings/Misc

Outbound SIP Registrations

Register Timeout

Register Attempts

Configure the register retry timeout (in seconds). The default setting is 20.

Configure the number of registration attempts before the UCM6100 gives up. The default setting is 0, which means the UCM6100 will keep trying until the server side accepts the registration request.

Video

Max Bit Rate (kb/s)

Support SIP Video

Generate Manager Events

Reject Non-Matching INVITE

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

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

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

If enabled, when rejecting an incoming INVITE or REGISTER request, the

UCM6100 will always reject with "401 Unauthorized" instead of notifying the requester whether there is a matching user or peer for the request.

This reduces the ability of an attacker to scan for valid SIP usernames.

The default setting is "No".

SIP SETTINGS/SESSION TIMER

Table 87: SIP Settings/Session Timer

Session Timers

Session Expire

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

The options are:

•

Originate

Always request and run session timer.

•

Accept

Run session timer only when requested by other UA.

•

Refuse

Do not run session timer.

Configure the maximum session refresh interval (in seconds). The default

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Min SE

Session Refresher setting is 1800.

Configure the minimum session refresh interval (in seconds). The default setting is 90.

Select the session refresher to be UAC or UAS. The default setting is

UAC.

SIP SETTINGS/TCP AND TLS

TCP Enable

TCP Bind Address

TLS Enable

TLS Bind Address

TLS Client Protocol

TLS Do Not Verify

TLS Self-Signed CA

TLS Cert

Table 88: SIP Settings/TCP and TLS

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

Configure the IP address for TCP server to bind to. 0.0.0.0 means binding to all interfaces. The port number is optional. If not specified, 5060 will be used.

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

Configure the IP address for TLS server to bind to. 0.0.0.0 means binding to all interfaces. The port number is optional. If not specified, 5061 will be

used.

Note:

The IP address must match the common name (hostname) in the certificate. Please do not bind a TLS socket to multiple IP addresses. For details on how to construct a certificate for SIP, please refer to the following document: http://tools.ietf.org/html/draft-ietf-sip-domain-certs

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

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

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

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


TLS CA Cert

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

Note:

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

This file must be named with the CA subject name hash value. It contains

CA's (Certificate Authority) public key, which is used to verify the accessed servers.

Note:

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

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

SIP SETTINGS/NAT

External IP Address

External Host

External Refresh

External TCP Port

External TLS Port

Local Network Address

Table 89: SIP Settings/NAT

Configure a static address and port (optional) that will be used in outbound SIP messages if the UCM6100 is behind NAT. If it's a hostname, it will only be looked up once.

Specify an external host name, which is similar to External Address except the host name will be looked up periodically based on the

"External Refresh" interval.

Configure the refresh interval for the external host (if used) The default setting is 10.

Configure the externally mapped TCP port when the UCM6100 is behind a static NAT or PAT.

Configures the externally mapped TLS port when UCM6100 is behind a static NAT or PAT.

Specify a list of network addresses that are considered inside of the NAT network. Multiple entries are allowed. If not configured, the external IP

address will not be set correctly.

A sample configuration could be as follows:

192.168.0.0/16

SIP SETTINGS/TOS

ToS For SIP


Table 90: SIP Settings/ToS

Configure the Type of Service for SIP packets. The default setting is


ToS For RTP Audio

ToS For RTP Video

Default Incoming/Outgoing

Registration Time

Max Registration/Subscription

Time

Min Registration/Subscription

Time

Music On Hold Interpret

Music On Hold Suggest

Enable Relaxed DTMF

DTMF Mode

RTP Timeout

RTP Hold Timeout

Trust Remote Party ID

Send Remote Party ID

Generate In-Band Ringing

None.

Configure the Type of Service for RTP audio packets. The default setting is None.

Configure the Type of Service for RTP video packets. The default setting is None.

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

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

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

Configure the Music On Hold class for the channel when being put on hold. This is used when the Music On Hold class is not set on the channel and the peer channel placing the call on hold doesn't have "Music On

Hold Suggest".

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

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

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

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

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

"RFC2833" will be used if offered, otherwise "Inband" will be used. The default setting is "RFC2833".

During an active call, if there is no RTP activity within the timeout (in seconds), the call will be terminated. The default setting is no timeout.

Note:

This setting doesn't apply to calls on hold.

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

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

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

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


Server User Agent

Send Compact SIP Headers

Add "user=phone" to URI

Progress and in-band audio.

•

No: The UCM6100 will send 180 Ringing if 183 Session Progress has not been sent yet. If audio path is established already with 183 then send in-band ringing.

•

Never: Whenever ringing occurs, the UCM6100 will send 180 Ringing as long as 200OK has not been set yet. Inband ringing will not be generated even the end point device is not working properly.

Configure the user agent string for the UCM6100.

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

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


VALUE-‐‑ADDED FEATURES

FAX SENDING

The UCM6100 supports sending Fax via web UI access. This feature can be found on web

UI->PBX->Value-added Features->Fax 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 UI. Users can also view the sending history is in the same web page.

Figure 147: Fax Sending in Web UI


ANNOUNCEMENTS CENTER

UCM6100 series supports Announcements Center feature starting from firmware version 1.0.9.26.

Announcements Center allows users to previously 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 148: Announcements Center

ANNOUNCEMENTS CENTER SETTINGS

Table 91: Announcements Center Settings

Name

Code

Custom Prompt

Ring Timeout

Configure a name for the newly created Announcements Center for 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.

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. The default value is 30 seconds.


GROUP SETTINGS

Table 92: Group Settings

Name Configure a name for the newly created group to identify the group.

Number

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 UI-> PBX->Value-added Features->

Announcements 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 149: Announcements Center Group Configuration

In this example, group “Test” has number 666. Extension 1000, 1001 and 1002 are in this group.

5. Click to create a new Announcement Center.


6. Give a name to the newly created Announcement Center.

7. Specify the code which will be used with group number to send the voice message to.

8. 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 150: 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 151: Announcements Center Example


STATUS AND REPORTING

PBX STATUS

The UCM6100 monitors the status for Trunks, Extensions, Queues, Conference Rooms, Interfaces and

Parking lot. It presents administrators the real time status in different sections under web

GUI->Status->PBX Status.

Figure 152: Status->PBX Status

TRUNKS

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

Status


Figure 153: Trunk Status

Table 93: Trunk Status

Display trunk status.

•

Analog trunk status:

Available

Busy

Unavailable

Unknown Error

•

SIP Peer trunk status:


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Unreachable: The hostname cannot be reached.

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

Reachable: The hostname can be reached.

•

SIP Register trunk status:

Registered

Unrecognized Trunk

Display trunk name Trunks

Type

Username

Display trunk Type:

•

Analog

•

SIP

•

IAX

Display username for this trunk.

Port/Hostname/IP Display Port for analog trunk, or Hostname/IP for VoIP (SIP/IAX) trunk.

Other operations are also available in trunk status section:

•

Click on "Trunks", the web page will redirect to trunk configuration page which can also be accessed via web GUI->PBX->Basic/Call Routes->Analog Trunks.

•

Click on to refresh the trunk status.

•

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

•

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

EXTENSIONS

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


Figure 154: Extension Status


Table 94: Extension Status

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 extension number.

First name and last name of the extension.

Extension

Name/Label

Message

Display message status for the extension.

Example: 2/4/1

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

Type

Displays extension type.

•

SIP User

•

IAX User

•

Analog User

•

Ring Groups

•

Voicemail Groups

Other operations are also available in extension status section:

•

Click on "Extensions", the web page will redirect to extension configuration page which can also be accessed via web GUI->PBX->Basic/Call Routes->Extensions.

•

Click on to refresh the extension status.

•

Click on one of the tabs the corresponding extensions accordingly.

•

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

•

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

to display

QUEUES

Users could see all the configured call queue status in this section. The following figure shows the call queue 6500 being in used.


Figure 155: Queue Status

The current call status (caller ID, duration), agent status, service level, calls summary

(completed/abandoned) are shown for the call queue. The agent status is defined as below.

Table 95: Agent Status

The agent is available/idle.

The agent is ringing.

The agent is talking/busy.

The agent has been logged out.

On the UCM6100, Service Level is defined as the percentage of high-quality calls over all calls in the call queue, where high-quality call means calls answered within 10 seconds.

Other operations are also available in queue status section:

•

Click on "Queues", the web page will redirect to call queue configuration page which can also be accessed via web GUI->PBX->Call Features->Call Queue.

•

Click on to refresh the call queue status.

•

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

•

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

CONFERENCE ROOMS

Users could see all the conference room status in this section. It shows all the configured conference rooms, current users, call duration for each user and conference call.


Figure 156: Conference Room Status


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Other operations are also available in conference room status section:

•

Click on "Conference Rooms", the web page will redirect to conference room configuration page which can also be accessed via web GUI->PBX->Call Features->Conference.

•

Click on to refresh the conference room status.

•

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

•

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

INTERFACES STATUS

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


Figure 157: UCM6116 Interfaces Status

Table 96: Interface Status Indicators

USB connected.

USB disconnected.

SD Card connected.

SD Card disconnected.

LAN/WAN connected.

LAN/WAN not configured.

LAN/WAN disconnected.

FXS/FXO connected.

FXS/FXO waiting.

FXS/FXO busy.

FXS/FXO not configured.


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FXS/FXO disconnected.

Other operations are also available in interface status section:

•

Click on "Interfaces Status", the web page will redirect to ports configuration page which can also be accessed via web GUI->PBX->Internal Options->Ports Config.

•

Click on to refresh the interface status.

•

Click on [ + ] to expand the interface details.

•

Click on [ - ] to hide the interface details.

PARKING LOT

The UCM6100 supports call park using feature code. When there is call being parked, this section will display the parking lot status.

Figure 158: Parking Lot Status

Table 97: Parking Lot Status

Caller ID

Channel

Extension

Timeout

Display the caller ID who parks the call.

Display channel for the call park.

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

Display timeout (in seconds) for the parked call. The status page will dynamically update this timer from 120 seconds (default) to 0. When the timer reaches 0, the caller who parks the call will be called back.

Other operations are also available in parking lot status section:

•

Click on "Parking Lot", the web page will redirect to feature codes page which can also be accessed via web GUI->PBX->Internal Options->Feature Codes.

•

Click on to refresh the parking lot status.

•

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

•

Click on [ - ] to hide the parking details.


SYSTEM STATUS

The UCM6100 system status can be accessed via Web GUI->Status->System Status, which displays the following system information.

•

General

•

Network

•

Storage Usage

•

Resource Usage

GENERAL

Under Web GUI->Status->System Status->General, users could check the hardware and software information for the UCM6100. Please see details in the following table.

Table 98: System Status->General

Status ->System Status -> General

Model Product model.

Part Number Product part number.

System Time

Up Time

Boot

Core

Base

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

System up time since the last reboot.

Boot version.

Core version.

Base version.

Program

Recovery

Program version. This is the main software release version.

Recovery version.

NETWORK

Under Web GUI->Status->System Status->Network, users could check the network information for the

UCM6100. Please see details in the following table.

Table 99: System Status->Network

Status -> System Status -> Network

MAC Address Global unique ID of device, in HEX format. The MAC address can be found on the


IP Address

Gateway

Subnet Mask

DNS Server

label coming with original box and on the label located on the bottom of the device.

IP address.

Default gateway address.

Subnet mask address.

DNS Server address.

STORAGE USAGE

Users could access the storage usage information from Web GUI->Status->System Status->Storage

Usage. It shows the available and used space for the following partitions.

•

Configuration partition

This partition contains PBX system configuration files and service configuration files.

•

Data partition

Voicemail, recording files, IVR file, Music on Hold files and etc.

•

USB disk

USB disk will display if connected.

•

SD Card

SD Card will display if connected.

Figure 159: System Status->Storage Usage

RESOURCE USAGE


When configuring and managing the UCM6100, users could access resource usage information to estimate the current usage and allocate the resources accordingly. Under Web GUI->Status->System

Status->Resource Usage, the current CPU usage and Memory usage are shown in the pie chart.

Figure 160: System Status->Resource Usage

SYSTEM EVENTS

The UCM6100 can monitor important system events, log the alerts and send Email notifications to the

system administrator.

ALERT EVENTS LIST

The system alert events list can be found under Web GUI->Status->System Events->Alert Events List.

The following event are currently supported on the UCM6100 which will have alert and/or Email generated if occurred:

Disk Usage

Modify Admin Password

Memory Usage

System Reboot

System Update

System Crash

Register SIP Failed

Register SIP Trunk Failed

Restore Config

User Login Success

User Login Failed

SIP Internal Call Failure


SIP Outgoing Call through Trunk Failure

Fail2ban Blocking

SIP Lost Registration

SIP Peer Trunk Status

Click on to configure the parameters for each event. See examples below.

1. Disk Usage.

Figure 161: System Events->Alert Events Lists: Disk Usage

•

Detect Cycle: The UCM6100 will perform the internal disk usage detection based on this cycle.

Users can enter the number and then select second(s)/minute(s)/hour(s)/day(s) to configure the cycle.

•

Alert Threshold: If the detected value exceeds the threshold (in percentage), the UCM6100 system will send the alert.

2. Memory Usage

Figure 162: System Events->Alert Events Lists: Memory Usage

•

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

•

Alert Threshold: If the detected value exceeds the threshold (in percentage), the UCM6100 system will send the alert.

3. System Reboot


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

•

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

4. System Crash

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

•

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

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

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

ALERT LOG

Under Web GUI->Status->System Events->Alert Log, system messages from triggered system events are listed as alert logs. The following screenshot shows system crash alert logs.


Figure 165: System Events->Alert 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

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 166: Filter for Alert Log


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ALERT CONTACT

Users could add administrator's Email address under Web GUI->Status->System Events->Alert Contact to send the alert notification to. Up to 10 Email addresses can be added.

CDR

A Call Detail Record (CDR) is a data record produced by telephone exchange activities or other telecommunications equipment documenting the details of a phone call that passed through the PBX. The

CDR is composed of the following data fields on the UCM6100.

•

Start Time. Format: 2013-03-27 16:47:03.

•

Call From. Format: "John Doe"<6012>.

•

Call To. Format: 6005.

•

Answered By. Format: 6005.

•

Call Time. Format: 0:00:10.

•

Talk Time. Format: 0:00:10

•

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

•

Options. Voice record playing/downloading/deleting.

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

Figure 167: CDR Filter

Inbound calls

Outbound calls

Internal calls

Table 100: CDR Filter Criteria

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

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

Internal calls are calls from one internal extension to another extension, which are


External calls

Inbound Trunks

Outbound Trunks

Start Time

End Time not sent over a trunk.

External calls are calls sent from one trunk to another trunk, which are not sent to any internal extension.

Select certain inbound trunk(s) and the CDR of calls going inbound through the trunk(s) will be filtered out.

Select certain outbound trunk(s) and the CDR of calls going outbound through the trunk(s) will be filtered out.

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.

Caller Number

Caller Name

Callee Number

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, case-insensitive, repeatable, only appears in the end.

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.

Figure 168: Call Report

Users could perform the following operations on the call report.

•

Sort

Click on the header of the column to sort by this category. For example, clicking on "Start Time" will


sort the report according to start time. Clicking on "Start Time" again will reverse the order.

•

Download 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 169: Call Report Entry with Audio Recording File

•

Automatic Download CDR Records

User could configure the UCM6100 to automatically download the CDR records and send the records to an Email address. Click on “Automatic Download Settings”, and configure the parameters in the dialog below.


Figure 170: Automatic Download Settings


To receive CDR record automatically from Email, check “Enable” and select a time period “By Day” “By

Week” or “By Month” for the automatic download period. Make sure you have entered an Email address to receive the CDR records.

DOWNLOADED CDR FILE

The downloaded CDR (.csv file) has different format from the web UI CDR. Here are some descriptions.

•

Call From, Call To

"Call From": the caller ID.

"Call To": the callee ID.

If "Call From" shows empty, "Call To" shows "s" (see highlight part in the picture below) and the "Source

Channel" contains "DAHDI", this means the call is from FXO/PSTN line. For FXO/PSTN line, we only know there is an incoming request when there is incoming call but we don't know the number being called. So

we are using "s" to match it where "s" means "start".

Figure 171: Downloaded CDR File Sample - Call To Shows "s"

•

Context

There are different context values that might show up in the downloaded CDR file. The actual value can vary case by case. Here are some sample values and their descriptions.

from-internal: internal extension makes outbound calls.

ext-did-XXXXX: inbound calls. It starts with "ext-did", and "XXXXX" content varies case by case, which

also relate to the order when the trunk is created.

ext-local: internal calls between local extensions.

•

Source Channel, Dest Channel

Sample 1:

Figure 172: Downloaded CDR File Sample - Source Channel and Dest Channel 1


DAHDI means it is an analog call, FXO or FXS.

For UCM6102, DAHDI/(1-2) are FXO ports, and DAHDI(3-4) are FXS ports.



For UCM6116, DAHDI/(1-16) are FXO ports, and DAHDI/(17-18) are FXS ports.

Sample 2:


"SIP" means it's a SIP call. There are three possible format:

(a) SIP/NUM-XXXXXX, where NUM is the local SIP extension number. The last XXXXX is a random string and can be ignored.

(c) SIP/trunk_X/NUM, where trunk_X is the internal trunk name, and NUM is the number to dial out through the trunk.

(c) SIP/trunk_X-XXXXXX, where trunk_X is the internal trunk name and it is an inbound call from this

trunk. The last XXXXX is a random string and can be ignored.

Sample 3:


This is a very special channel name. If it shows up, most likely it means a conference call.

There are some other possible values, but these values are almost the application name which are used

by the dialplan.

IAX2/NUM-XXXXXXX: it means this is an IAX call.

Local/@from-internal-XXXXX: it is used internally to do some special feature procedure. We can simply ignore it.

Hangup: the call is hung up from the dialplan. This indicates there are some errors or it has run into abnormal cases.

Playback: play some prompts to you, such as 183 response or run into an IVR.

ReadExten: collect numbers from user. It may occur when you input PIN codes or run into DISA

STATISTICS


CDR Statistics is an additional feature on the UCM6100 which provides users a visual overview of the call report across the time frame. Users can filter with different criteria to generate the statistics chart.

Trunk Type

Call Type

Time Range

Figure 175: CDR Statistics

Table 101: CDR Statistics Filter Criteria

Select one of the following trunk type.

•

All

•

SIP Calls

•

PSTN Calls

Select one or more in the following checkboxes.

•

Inbound calls

•

Outbound calls

•

Internal calls

•

External calls

•

All calls

•

By month (of the selected year).

•

By week (of the selected year).

•

By day (of the specified month for the year).

•

By hour (of the specified date).

•

By range. For example, 2013-01 To 2013-03.

RECORDING FILES

This page lists all the recording files recorded by "Auto Record" per extension/ring group/call queue/trunk, or via feature code "Audio Mix Record". If external storage device is plugged in, for example, SD card or

USB drive, the files are stored on the external storage. Otherwise, internal storage will be used on the

UCM6100.


Figure 176: CDR->Recording Files

•

Click on “Delete Selected Recording Files” to delete the recording files.

•

Click on “Delete All Recording Files” to delete all recording files.

•

Click on to download the recording file in .wav format.

•

Click on to delete the recording file.

•

To sort the recording file, click on the title "Caller", "Callee" or "Call Time" for the corresponding column. Click on the title again can switch the sorting mode between ascending order or descending

order.

API CONFIGURATION

The UCM6100 supports third party billing interface API for external billing software to access CDR and call recordings on the PBX. The API uses HTTPS to request the CDR data and call recording data matching given parameters as configured on the third party application.

Before accessing the API, the administrators need enable API and configure the access/authentication

information on the UCM6100 first. The API configuration parameters are listed in the table below.

Table 102: API Configuration Files

Enable Enable/Disable API. The default setting is disabled.

TLS Bind Address Configure the IP address for TLS server to bind to. "0.0.0.0" means binding to all interfaces. The port number is optional and the default port number is 8443. The IP address must match the common name (host name) in the certificate so that the TLS socket won't bind to multiple IP addresses. The default setting is 0.0.0.0:8443.

TLS Private Key Upload TLS private key. The size of the key file must be under 2MB. This file will be renamed as 'private.pem' automatically.

TLS Cert Upload TLS cert. The size of the certificate must be under 2MB. This is the certificate file (*.pem format only) for TLS connection. This file will be renamed as

"certificate.pem" automatically. It contains private key for the client and signed


TLS Authentication

Name

TLS Authentication

Password

Permitted certificate for the server.

Configure the user name for TLS authentication. If not configured, authentication will be skipped.

Configure the password for TLS authentication. This is optional.

Specify a list of IP addresses permitted by API. This creates an AIP-specific access control list. Multiple entries are allowed.

For example, "192.168.40.3/255.255.255.255" denies access from all IP addresses except 192.168.40.3.

The default setting is blank, meaning all IPs will be denied. Users must set permitted

IP address before connecting to the API.

For more details on CDR API (Access to Call Detail Records) and REC API (Access to Call Recording

Files), please refer the document in the link here:

http://www.grandstream.com/products/ucm_series/ucm61xx/documents/ucm61xx_cdr_rec_api_guide.pdf


UPGRADING AND MAINTENANCE

UPGRADING

The UCM6100 can be upgraded to a new firmware version remotely or locally. This section describes how to upgrade your UCM6100 via network or local upload.

UPGRADING VIA NETWORK

The UCM6100 can be upgraded via TFTP/HTTP/HTTPS by configuring the URL/IP Address for the

TFTP/HTTP/HTTPS server and selecting a download method. Configure a valid URL for TFTP, HTTP or

HTTPS;; the server name can be FQDN or IP address.

Examples of valid URLs:

firmware.grandstream.com

The upgrading configuration can be accessed via Web GUI->Maintenance->Upgrade.

Upgrade Via

Firmware Server Path

Firmware File Prefix

Firmware File Suffix


Figure 177: Network Upgrade

Table 103: Network Upgrade Configuration

Allow users to choose the firmware upgrade method: TFTP, HTTP or

HTTPS.

Define the server path for the firmware server.

If configured, only the firmware with the matching encrypted prefix will be downloaded and flashed into the UCM6100.

If configured, only the firmware with the matching encrypted postfix will be


HTTP/HTTPS User Name

HTTP/HTTPS Password downloaded and flashed into the UCM6100.

The user name for the HTTP/HTTPS server.

The password for the HTTP/HTTPS server.

Please follow the steps below to upgrade the firmware remotely.

•

Enter the firmware server path under Web GUI->Maintenance->Upgrade.

•

Click on "Save". Then reboot the device to start the upgrading process.

•

Please be patient during the upgrading process. Once done, a reboot message will be displayed in the

LCD.

•

Manually reboot the UCM6100 when it's appropriate to avoid immediate service interruption. After it boots up, log in the web GUI to check the firmware version.

UPGRADING VIA LOCAL UPLOAD

If there is no HTTP/TFTP server, users could also upload the firmware to the UCM6100 directly via Web

GUI. Please follow the steps below to upload firmware locally.

•

Download the latest UCM6100 firmware file from the following link and save it in your PC. http://www.grandstream.com/support/firmware

•

Log in the Web GUI as administrator in the PC.

•

Go to Web GUI->Maintenance->Upgrade, upload the firmware file by clicking on the firmware file from your PC. The default firmware file name is ucm6100fw.bin

and select

•

Click on to start upgrading.

Figure 178: Local Upgrade


Figure 179: Upgrading Firmware Files

•

Wait until the upgrading process is successful and a window will be popped up in the Web GUI.

Figure 180: Reboot UCM6100

•

Click on "OK" to reboot the UCM6100 and check the firmware version after it boots up.

--------------------------------------------------------------------------------------------------------------------------------------------

Note:

Please do not interrupt or power cycle the UCM6100 during upgrading process.

--------------------------------------------------------------------------------------------------------------------------------------------


NO LOCAL FIRMWARE SERVERS

For users that would like to use remote upgrading without a local TFTP server, Grandstream offers a

NAT-friendly HTTP server. This enables users to download the latest software upgrades for their devices via this server. Please refer to the webpage: http://www.grandstream.com/support/firmware .

Alternatively, users can download a free TFTP or HTTP server and conduct a local firmware upgrade. A free windows version TFTP server is available for download from : http://www.solarwinds.com/products/freetools/free_tftp_server.aspx

http://tftpd32.jounin.net

Instructions for local firmware upgrade via TFTP:

1. Unzip the firmware files and put all of them in the root directory of the TFTP server;;

2. Connect the PC running the TFTP server and the UCM6100 to the same LAN segment;;

3. Launch the TFTP server and go to the File menu->Configure->Security to change the TFTP server's default setting from "Receive Only" to "Transmit Only" for the firmware upgrade;;

4. Start the TFTP server and configure the TFTP server in the UCM6100 web configuration interface;;

5. Configure the Firmware Server Path to the IP address of the PC;;

6. Update the changes and reboot the UCM6100.

End users can also choose to download a free HTTP server from http://httpd.apache.org/ or use

Microsoft IIS web server.

BACKUP

The UCM6100 configuration can be backed up locally or via network. The backup file will be used to restore the configuration on UCM6100 when necessary.

BACKUP/RESTORE

Users could backup the UCM6100 configurations for restore purpose under Web

GUI->Maintenance->Backup->Local Backup.

Click on to create a new backup file. Then the following dialog will show.


Figure 181: 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.

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.

Figure 182: Backup / Restore

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.

Figure 183: Local Backup

DATA SYNC

Besides local backup, users could backup the voice records/voice mails/CDR/FAX in a daily basis to a remote server via SFTP protocol automatically under Web GUI->Maintenance->Backup->Data Sync.


Figure 184: Data Sync

Table 104: Data Sync Configuration


Page 293 of 304

Enable Backup

Account

Password

Enable the auto backup function. The default setting is "No".

Enter the Account name on the SFTP backup server.

Enter the Password associate with the Account on the SFTP backup server.

Server Address

Backup Time

Enter the SFTP server address.

Enter 0-23 to specify the backup hour of the day.

Before saving the configuration, users could click on "Test Connection". The UCM6100 will then try connecting the server to make sure the server is up and accessible for the UCM6100. Save the changes and all the backup logs will be listed on the web page. After data sync is configured, users could also

instead of waiting for the backup manually synchronize all data by clicking on

time 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->Maintenance->Backup->Local Backup.

•

A list of previous configuration backups is displayed on the web page. Users could click on of the desired backup file and it will be restored to the UCM6100.

•

If users have other backup files on PC to restore on the UCM6100, click on "Upload Backup File" first and select it from local PC to upload on the UCM6100. Once the uploading is done, this backup file will be displayed in the list of previous configuration backups for restore purpose. Click on from the backup file.

to restore


Figure 185: Restore UCM6100 from Backup File

--------------------------------------------------------------------------------------------------------------------------------------------

Note:

• The uploaded backup file must be a tar file with no special characters like *,!,#,@,&,$,%,^,(,),/,\,space in the file name.

• The uploaded back file size must be under 10MB.

--------------------------------------------------------------------------------------------------------------------------------------------

CLEANER

Users could configure to clean the Call Detail Report/Voice Records/Voice Mails/FAX automatically under

Web GUI->Maintenance->Cleaner.


Figure 186: Cleaner

Table 105: Cleaner Configuration

Enable CDR Cleaner

CDR Clean Time

Clean Interval

Enable VR Cleaner

Enable the CDR Cleaner function.

Enter 0-23 to specify the hour of the day to clean up CDR.

Enter 1-30 to specify the day of the month to clean up CDR.

Choose Cleaner File

VR Clean Threshold

Enter the Voice Records Cleaner function.

Select the files for system automatic clean.

•

Recording Files

•

Conference

•

Queue

•

Voicemail

•

Fax

Specify the Voice Records threshold from 0 to 99 by using local storage status in percentage.

VR Clean Time

Clean Interval

Enter 0-23 to specify the hour of the day to clean up Voice Records.

Enter 1-30 to specify the day of the month to clean up Voice Records.

All the cleaner logs will be listed on the bottom of the page.


--------------------------------------------------------------------------------------------------------------------------------------------

Note:

Cleaner will delete data based on Recording Storage selection. If USB Disk is selected, Cleaner will only clean data in USB and local data will leave untouched. If Enable auto change is selected and USB disk is connected, Cleaner will only delete data in USB drive. Recordings Storage function can be found under web UI-> Settings-> Recordings Storage-> Recordings Storage.

--------------------------------------------------------------------------------------------------------------------------------------------

RESET AND REBOOT

Users could perform reset and reboot under Web GUI->Maintenance->Reset and Reboot.

To factory reset the device, select the mode type first. There are two different types for reset.

• User Data: All the data including voicemail, recordings, IVR Prompt, Music on Hold, CDR and backup files will be cleared.

•

All: All the configurations and data will be reset to factory default.

Figure 187: Reset and Reboot

SYSLOG

On the UCM6100, users could dump the syslog information to a remote server under Web

GUI->Maintenance->Syslog. Enter the syslog server hostname or IP address and select the module/level for the syslog information.


The default syslog level for all modules is "error", which is recommended in your UCM6100 settings because it can be helpful to locate the issues when errors happen.

Some typical modules for UCM6100 functions are as follows and users can turn on "notic" and "verb"

levels besides "error" level. pbx: This module is related to general PBX functions. chan_sip: This module is related to SIP calls. chan_dahdi: This module is related to analog calls (FXO/FXS).

app_meetme: This module is related to conference bridge.

--------------------------------------------------------------------------------------------------------------------------------------------

Note:

Syslog is usually for debugging and troubleshooting purpose. Turning on all levels for all syslog modules is not recommended for daily usage. Too many syslog print might cause traffic and affect system performance.

--------------------------------------------------------------------------------------------------------------------------------------------

TROUBLESHOOTING

On the UCM6100, users could capture traces, ping remote host and traceroute remote host for

troubleshooting purpose under Web GUI->Maintenance->Troubleshooting.

ETHERNET CAPTURE

The captured trace can be downloaded for analysis. Also the instructions or result will be displayed in the web GUI output result.


Figure 188: Ethernet Capture


Page 298 of 304

The output result is in .pcap format. Therefore, users could specify the capture filter as used in general network traffic capture tool (host, src, dst, net, protocol, port, port range) before starting capturing the

trace.

IP PING

Enter the target host in host name or IP address. Then press "Start" button. The output result will dynamically display in the window below.

Figure 189: 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 190: Traceroute

ANALOG RECORD TRACE

Analog record trace can be used to troubleshoot analog trunk issue, for example, the UCM6100 user has caller ID issue for incoming call from Analog trunk. Users can access analog record trance under web

GUI->Maintenance->Troubleshooting ->Analog Record Trace.

Here is the step to capture trace:

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 191: Troubleshooting Analog Trunks

After capturing the trace, users can download it for basic analysis. Or you can contact Grandstream

Technical support in the following link for further assistance if the issue is not resolved. http://www.grandstream.com/index.php/support

SERVICE CHECK

Enable Service Check to periodically check 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.

Figure 192: Service Check

REMOTE ACCESS

SSH ACCESS

SSH switch now is available via web UI and LCD. User can enable or disable SSH access directly from web UI or LCD screen. For web SSH access, please log in UCM 6100 web interface and go to


Maintenance->Remote Access->SSH Access. By default, SSH access is disabled for security concerns.

It is highly recommend to only enable SSH access for debugging purpose.

Figure 193: SSH Access



EXPERIENCING THE UCM6100 SERIES IP PBX

Please visit our website: http://www.grandstream.com

to receive the most up- to-date updates on firmware releases, additional features, FAQs, documentation and news on new products.

We encourage you to browse our product related documentation , FAQs and User and Developer Forum for answers to your general questions. If you have purchased our products through a Grandstream

Certified Partner or Reseller, please contact them directly for immediate support.

Our technical support staff is trained and ready to answer all of your questions. Contact a technical support member or submit a trouble ticket online to receive in-depth support.

Thank you again for purchasing Grandstream UCM6100 series IP PBX appliance, it will be sure to bring

convenience and color to both your business and personal life.

* Asterisk is a Registered Trademark of Digium, Inc.


FCC Caution:

Any Changes or modifications not expressly approved by the party responsible for compliance could void the user's authority to operate the equipment.

This device complies with part 15 of the FCC Rules. Operation is subject to the following two conditions:

(1) This device may not cause harmful interference, and (2) this device must accept any interference

received, including interference that may cause undesired operation.

Note: This equipment has been tested and found to comply with the limits for a Class B digital device, pursuant to part 15 of the FCC Rules. These limits are designed to provide reasonable protection against harmful interference in a residential installation. This equipment generates, uses and can radiate radio frequency energy and, if not installed and used in accordance with the instructions, may cause harmful interference to radio communications. However, there is no guarantee that interference will not occur in a particular installation. If this equipment does cause harmful interference to radio or television reception, which can be determined by turning the equipment off and on, the user is encouraged to try to correct the interference by one or more of the following measures:

- Reorient or relocate the receiving antenna.

- Increase the separation between the equipment and receiver.

- Connect the equipment into an outlet on a circuit different from that to which the receiver is connected.

- Consult the dealer or an experienced radio/TV technician for help.

Regulatory Information

U.S. FCC Part 68 Statement

This equipment complies with Part 68 of the FCC rules and the requirements adopted by the ACTA. The unit bears a label on the back which contains among other information a product identifier in the format

US: GNIIS00BUCM6104/US: GNIIS00BUCM6116. If requested, this number must be provided to the telephone company.

This equipment uses the following standard jack types for network connection: RJ11C.

This equipment contains an FCC compliant modular jack. It is designed to be connected to the telephone network or premises wiring using compatible modular plugs and cabling which comply with the requirements of FCC Part 68 rules.

The Ringer Equivalence Number, or REN, is used to determine the number of devices which may be connected to the telephone line. An excessive REN may cause the equipment to not ring in response to an incoming call. In most areas, the sum of the RENs of all equipment on a line should not exceed five (5.0).

In the unlikely event that this equipment causes harm to the telephone network, the telephone company can temporarily disconnect your service. The telephone company will try to warn you in advance of any such disconnection, but if advance notice isn't practical, it may disconnect the service first and notify you as soon as possible afterwards. In the event such a disconnection is deemed necessary, you will be advised of your right to file a complaint with the FCC.

From time to time, the telephone company may make changes in its facilities, equipment, or operations which could affect the operation of this equipment. If this occurs, the telephone company is required to provide you with advance notice so you can make the modifications necessary to obtain uninterrupted service.

There are no user serviceable components within this equipment. See Warranty flyer for repair or warranty information.

It shall be unlawful for any person within the United States to use a computer or other electronic device to send any message via a telephone facsimile unless such message clearly contains, in a margin at the top or bottom of each transmitted page or on the first page of the transmission, the date and time it is sent and an identification of the business, other entity, or individual sending the message and the telephone number of the sending machine or of such business, other entity, or individual. The telephone number provided may not be a 900 number or any other number for which charges exceed local or long distance transmission charges. Telephone facsimile machines manufactured on and after December 20, 1992, must clearly mark such identifying information on each transmitted message. Facsimile modem boards manufactured on and after December 13, 1995, must comply with the requirements of this section.

This equipment cannot be used on public coin phone service provided by the telephone company.

Connection to Party Line Service is subject to state tariffs. Contact your state public utility commission, public service commission, or corporation commission for more information.

Grandstream Networks UCM6100 Series User manual

UCM6100 Series IP PBX

User Manual

UCM6100 Series IP PBX User Manual

Table of Content

GNU GPL INFORMATION

CHANGE LOG

FIRMWARE VERSION 1.0.9.26

FIRMWARE VERSION 1.0.9.25

FIRMWARE VERSION 1.0.9.23

FIRMWARE VERSION 1.0.8.12

FIRMWARE VERSION 1.0.7.11

FIRMWARE VERSION 1.0.6.11

FIRMWARE VERSION 1.0.6.10

FIRMWARE VERSION 1.0.5.19

FIRMWARE VERSION 1.0.5.14

FIRMWARE VERSION 1.0.4.7

FIRMWARE VERSION 1.0.3.13

FIRMWARE VERSION 1.0.2.21

FIRMWARE VERSION 1.0.1.22

WELCOME

PRODUCT OVERVIEW

FEATURE HIGHTLIGHTS

TECHNICAL SPECIFICATIONS

Signaling and Control

Interfaces

Voice/Video Capabilities

Security

Physical

Additional Features

INSTALLATION

EQUIPMENT PACKAGING

CONNECT YOUR UCM6100

SAFETY COMPLIANCES

WARRANTY

GETTING STARTED

USE THE LCD MENU

USE THE LED INDICATORS

USE THE WEB GUI

MAKE YOUR FIRST CALL

SYSTEM SETTINGS

USER MANAGEMENT

NETWORK SETTINGS

FIREWALL

CHANGE PASSWORD

LDAP SERVER

HTTP SERVER

EMAIL SETTINGS

TIME SETTINGS

NTP SERVER

RECORDINGS STORAGE

LOGIN TIMEOUT SETTINGS

PROVISIONING

OVERVIEW

CONFIGURATION ARCHITECTURE FOR END POINT DEVICE

AUTO PROVISIONING SETTINGS

DISCOVERY

GLOBAL CONFIGURATION

MODEL CONFIGURATION

DEVICE CONFIGURATION

SAMPLE APPLICATION

EXTENSIONS

CREATE NEW USER

BATCH ADD EXTENSIONS

SEARCH AND EDIT EXTENSION

EXPORT EXTENSIONS

IMPORT EXTENSIONS

EMAIL TO USER

TRUNKS

ANALOG TRUNKS

VOIP TRUNKS

SLA STATION

CREATE/EDIT SLA STATION

SAMPLE CONFIGURATION

CALL ROUTES

OUTBOUND ROUTES

INBOUND ROUTES

CONFERENCE BRIDGE

IVR

CONFIGURE IVR

SAMPLE CONFIGURATION FOR FAX-‐‑TO-‐‑EMAIL

BUSY CAMP-‐‑ON

ONE-‐‑KEY DIAL