AudioCodes MediaPack MP-102, MediaPack MP-104, MediaPack MP-108, MP-124, MediaPack MP-124 User Manual

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MP-1xx SIP User’s Manual
Version 4.4
Document #: LTRT-65404
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Notice
© Copyright 2005 AudioCodes Ltd. All rights reserved.
This document is subject to change without notice.
Date Published: Mar-01-2005 Date Printed: Mar-16-2005
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MP-1xx SIP User’s Manual Contents
Version 4.4 3 March 2005
Table of Contents
1 Overview ....................................................................................................................15
1.1 Introduction.....................................................................................................................................15
1.2 Gateway Description.......................................................................................................................15
1.3 SIP Overview..................................................................................................................................17
1.4 MP-1xx Features ............................................................................................................................18
1.4.1 General Features ....................................................................................................................18
1.4.2 Hardware Features .................................................................................................................18
1.4.3 SIP Features ...........................................................................................................................18
2 MP-1xx Physical Description....................................................................................21
2.1 MP-1xx Front Panel........................................................................................................................21
2.1.1 MP-1xx Front Panel Buttons...................................................................................................21
2.1.2 MP-1xx Front Panel LEDs ......................................................................................................22
2.2 MP-1xx Rear Panel.........................................................................................................................22
2.2.1 MP-10x Rear Panel.................................................................................................................22
2.2.2 MP-124 Rear Panel ................................................................................................................23
3 Installing the MP-1xx.................................................................................................25
3.1 Unpacking.......................................................................................................................................25
3.2 Package Contents ..........................................................................................................................25
3.3 Mounting the MP-1xx......................................................................................................................26
3.3.1 Mounting the MP-1xx on a Desktop........................................................................................26
3.3.2 Installing the MP-10x in a 19-inch Rack .................................................................................26
3.3.3 Installing the MP-124 in a 19-inch Rack .................................................................................27
3.3.4 Mounting the MP-10x on a Wall..............................................................................................28
3.4 Cabling the MP-1xx ........................................................................................................................29
3.4.1 Connecting the MP-1xx RS-232 Port to Your PC...................................................................32
3.4.1.1 Configuring the Serial Connection................................................................................32
3.4.2 Cabling the Lifeline Phone......................................................................................................32
4 Getting Started...........................................................................................................35
4.1 Assigning the MP-1xx IP Address ..................................................................................................35
4.1.1 Assigning an IP Address Using HTTP ....................................................................................35
4.1.2 Assigning an IP Address Using BootP....................................................................................36
4.2 Restoring Networking Parameters to their Initial State...................................................................36
4.3 Configure the MP-1xx Basic Parameters .......................................................................................37
5 Configuring the MP-1xx............................................................................................39
5.1 Configuration Concepts ..................................................................................................................39
5.2 Overview of the Embedded Web Server ........................................................................................39
5.3 Computer Requirements.................................................................................................................39
5.4 Password Control ...........................................................................................................................40
5.4.1 Embedded Web Server Username & Password.....................................................................40
5.5 Configuring the Web Interface via the ini File.................................................................................40
5.5.1 Limiting the Embedded Web Server to Read-Only Mode.......................................................40
5.5.2 Disabling the Embedded Web Server.....................................................................................40
5.6 Accessing the Embedded Web Server...........................................................................................41
5.6.1 Using Internet Explorer to Access the Embedded Web Server..............................................41
5.7 Getting Acquainted with the Web Interface ....................................................................................42
5.7.1 Main Menu Bar........................................................................................................................42
5.7.2 Saving Changes......................................................................................................................43
5.7.3 Entering Phone Numbers in Various Tables...........................................................................43
5.8 Protocol Management.....................................................................................................................44
5.8.1 Protocol Definition Parameters ...............................................................................................44
5.8.1.1 General Parameters .....................................................................................................44
5.8.1.2 Proxy & Registration Parameters .................................................................................48
5.8.1.3 Coders ..........................................................................................................................53
5.8.1.4 DTMF & Dialing Parameters.........................................................................................55
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5.8.2 Configuring the Advanced Parameters...................................................................................58
5.8.2.1 General Parameters .....................................................................................................58
5.8.2.2 Supplementary Services...............................................................................................63
5.8.2.3 Keypad Features ..........................................................................................................66
5.8.3 Configuring the Number Manipulation Tables ........................................................................68
5.8.3.1 Dialing Plan Notation ....................................................................................................71
5.8.4 Configuring the Routing Tables ..............................................................................................73
5.8.4.1 General Parameters .....................................................................................................73
5.8.4.2 Tel to IP Routing Table.................................................................................................75
5.8.4.3 IP to Hunt Group Routing .............................................................................................78
5.8.4.4 Internal DNS Table .......................................................................................................80
5.8.4.5 Reasons for Alternative Routing...................................................................................81
5.8.5 Configuring the Profile Definitions ..........................................................................................83
5.8.5.1 Coder Group Settings...................................................................................................83
5.8.5.2 Tel Profile Settings........................................................................................................85
5.8.5.3 IP Profile Settings .........................................................................................................87
5.8.6 Configuring the Endpoint Phone Numbers .............................................................................89
5.8.7 Configuring the Hunt Group Settings......................................................................................91
5.8.8 Configuring the Endpoint Settings ..........................................................................................93
5.8.8.1 Authentication ...............................................................................................................93
5.8.8.2 Automatic Dialing..........................................................................................................94
5.8.8.3 Caller ID........................................................................................................................95
5.8.8.4 Call Forward .................................................................................................................96
5.8.8.5 Caller ID Permissions ...................................................................................................98
5.8.9 Configuring the FXO Parameters............................................................................................99
5.8.10 Protocol Management ini File Parameters............................................................................101
5.9 Advanced Configuration ...............................................................................................................103
5.9.1 Configuring the Network Settings .........................................................................................103
5.9.1.1 Configuring the SNMP Managers Table.....................................................................107
5.9.1.2 Multiple Routers Support ............................................................................................112
5.9.1.3 Simple Network Time Protocol Support......................................................................112
5.9.2 Configuring the Channel Settings .........................................................................................113
5.9.2.1 Dynamic Jitter Buffer Operation .................................................................................120
5.9.3 Restoring and Backing up the Gateway Configuration.........................................................121
5.9.4 Regional Settings..................................................................................................................122
5.9.5 Changing the MP-1xx Username and Password..................................................................123
5.10 Status & Diagnostics.....................................................................................................................124
5.10.1 Gateway Statistics ................................................................................................................124
5.10.1.1 IP Connectivity............................................................................................................124
5.10.1.2 Call Counters ..............................................................................................................125
5.10.2 Monitoring the MP-1xx Channels..........................................................................................127
5.10.3 Activating the Internal Syslog Viewer ...................................................................................129
5.10.4 System Information ...............................................................................................................130
5.11 Software Update...........................................................................................................................131
5.11.1 Software Upgrade Wizard.....................................................................................................131
5.11.2 Auxiliary Files........................................................................................................................137
5.11.2.1 Loading the Auxiliary Files via the ini File...................................................................138
5.12 Save Configuration .......................................................................................................................139
5.13 Resetting the MP-1xx ...................................................................................................................140
6 ini File Configuration of the MP-1xx ......................................................................141
6.1 Secured ini File.............................................................................................................................141
6.2 Modifying an ini File......................................................................................................................141
6.3 The ini File Structure.....................................................................................................................142
6.3.1 The ini File Structure Rules...................................................................................................142
6.3.2 The ini File Example .............................................................................................................142
7 Configuration Files..................................................................................................143
7.1 Configuring the Call Progress Tones and Distinctive Ringing File...............................................143
7.1.1 Format of the Call Progress Tones Section in the ini File ....................................................143
7.1.2 Format of the Distinctive Ringing Section in the ini File .......................................................145
7.1.2.1 Examples of Various Ringing Signals.........................................................................146
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7.2 Prerecorded Tones (PRT) File .....................................................................................................147
7.2.1 PRT File Format....................................................................................................................147
7.3 The Coefficient Configuration File ................................................................................................148
8 Gateway Capabilities Description..........................................................................149
8.1 Proxy or Registrar Registration Example .....................................................................................149
8.2 Configuring the DTMF Transport Types.......................................................................................150
8.3 Configuring the Gateway’s Alternative Routing (based on Connectivity and QoS)......................152
8.3.1 Alternative Routing Mechanism ............................................................................................152
8.3.2 Determining the Availability of Destination IP Addresses.....................................................152
8.3.3 Relevant Parameters ............................................................................................................153
8.4 Working with Supplementary Services.........................................................................................153
8.4.1 Call Hold and Retrieve..........................................................................................................153
8.4.1.1 Initiating Hold/Retrieve................................................................................................153
8.4.1.2 Receiving Hold / Retrieve ...........................................................................................154
8.4.2 Consultation / Alternate.........................................................................................................154
8.4.3 Call Transfer..........................................................................................................................154
8.4.4 Call Forward..........................................................................................................................155
8.4.5 Call Waiting...........................................................................................................................155
8.4.6 Message Waiting Indication..................................................................................................156
8.5 Call Termination on MP-1xx/FXO.................................................................................................156
8.6 Mapping PSTN Release Cause to SIP Response .......................................................................157
8.7 Call Detail Report..........................................................................................................................158
8.8 Metering Tones Relay...................................................................................................................159
8.9 Configuration Examples................................................................................................................160
8.9.1 Establishing a Call between Two Gateways.........................................................................160
8.9.2 SIP Call Flow.........................................................................................................................161
8.9.3 SIP Authentication Example .................................................................................................163
8.9.4 Remote IP Extension between FXO and FXS......................................................................165
8.9.4.1 Dialing from Remote Extension ..................................................................................166
8.9.4.2 Dialing from other PBX line, or from PSTN ................................................................166
8.9.4.3 MP-108/FXS Configuration (using the Embedded Web Server)................................167
8.9.4.4 MP-108/FXO Configuration (using the Embedded Web Server)................................168
9 Diagnostics..............................................................................................................169
9.1 MP-1xx Self-Testing .....................................................................................................................169
9.1.1 Rapid Self-Test Mode ...........................................................................................................169
9.1.2 Detailed Self-Test Mode .......................................................................................................169
9.2 Troubleshooting the MP-1xx via the RS-232 Port........................................................................170
9.2.1 Viewing the Gateway’s Information ......................................................................................170
9.2.2 Changing the Networking Parameters..................................................................................170
9.3 Syslog Support .............................................................................................................................171
9.3.1 Syslog Servers......................................................................................................................171
9.3.2 Operation ..............................................................................................................................171
9.3.2.1 Sending the Syslog Messages ...................................................................................171
9.3.2.2 Setting the Syslog Server ...........................................................................................171
9.3.2.3 The ini File Example for Syslog ..................................................................................172
9.4 Solutions to Possible Problems....................................................................................................172
9.4.1 General .................................................................................................................................172
10 BootP/DHCP Support..............................................................................................173
10.1 Startup Process ............................................................................................................................173
10.2 DHCP Support ..............................................................................................................................175
10.3 BootP Support ..............................................................................................................................176
10.3.1 Upgrading the MP-1xx ..........................................................................................................176
10.3.2 Vendor Specific Information Field.........................................................................................176
11 SNMP-Based Management .....................................................................................179
11.1 About SNMP .................................................................................................................................179
11.1.1 SNMP Message Standard ....................................................................................................179
11.1.2 SNMP MIB Objects ...............................................................................................................180
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11.1.3 SNMP Extensibility Feature ..................................................................................................180
11.2 Carrier Grade Alarm System ........................................................................................................181
11.2.1 Active Alarm Table................................................................................................................181
11.2.2 Alarm History.........................................................................................................................181
11.3 Cold Start Trap .............................................................................................................................181
11.4 Third-Party Performance Monitoring Measurements ...................................................................182
11.5 Supported MIBs............................................................................................................................182
11.6 SNMP Interface Details ................................................................................................................185
11.6.1 SNMP Community Names....................................................................................................185
11.6.1.1 Configuration of Community Strings via the ini File....................................................185
11.6.1.2 Configuration of Community Strings via SNMP..........................................................185
11.6.2 Trusted Managers.................................................................................................................186
11.6.2.1 Configuration of Trusted Managers via ini File...........................................................186
11.6.2.2 Configuration of Trusted Managers via SNMP ...........................................................186
11.6.3 SNMP Ports ..........................................................................................................................187
11.6.4 Multiple SNMP Trap Destinations.........................................................................................188
11.6.4.1 Configuration via the ini File .......................................................................................188
11.6.4.2 Configuration via SNMP .............................................................................................189
11.7 SNMP Manager Backward Compatibility......................................................................................190
11.8 AudioCodes’ Element Management System................................................................................190
12 Selected Technical Specifications.........................................................................191
Appendix A MP-1xx SIP Software Kit.......................................................................195
Appendix B The BootP/TFTP Configuration Utility.................................................197
B.1 When to Use the BootP/TFTP ......................................................................................................197
B.2 An Overview of BootP...................................................................................................................197
B.3 Key Features ................................................................................................................................197
B.4 Specifications................................................................................................................................198
B.5 Installation.....................................................................................................................................198
B.6 Loading the cmp File, Booting the Device....................................................................................198
B.7 BootP/TFTP Application User Interface........................................................................................199
B.8 Function Buttons on the Main Screen ..........................................................................................199
B.9 Log Window ..................................................................................................................................200
B.10 Setting the Preferences ................................................................................................................201
B.10.1 BootP Preferences ................................................................................................................201
B.10.2 TFTP Preferences.................................................................................................................202
B.11 Configuring the BootP Clients ......................................................................................................203
B.11.1 Adding Clients .......................................................................................................................203
B.11.2 Deleting Clients .....................................................................................................................204
B.11.3 Editing Client Parameters .....................................................................................................204
B.11.4 Testing the Client ..................................................................................................................204
B.11.5 Setting Client Parameters .....................................................................................................205
B.11.6 Using Command Line Switches ............................................................................................206
B.12 Managing Client Templates..........................................................................................................207
Appendix C RTP/RTCP Payload Types and Port Allocation..................................209
C.1 Packet Types Defined in RFC 1890 .............................................................................................209
C.2 Defined Payload Types.................................................................................................................209
C.3 Default RTP/RTCP/T.38 Port Allocation.......................................................................................210
Appendix D Fax & Modem Transport Modes ..........................................................211
D.1 Fax/Modem Settings.....................................................................................................................211
D.2 Configuring Fax Relay Mode........................................................................................................211
D.3 Configuring Fax/Modem Bypass Mode ........................................................................................211
D.4 Supporting V.34 Faxes.................................................................................................................212
D.4.1 Using Bypass Mechanism for V.34 Fax Transmission .........................................................212
D.4.2 Using Relay mode for both T.30 and V.34 faxes..................................................................212
Appendix E Customizing the MP-1xx Web Interface..............................................213
E.1 Replacing the Main Corporate Logo.............................................................................................213
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E.1.1 Replacing the Main Corporate Logo with an Image File.......................................................213
E.1.2 Replacing the Main Corporate Logo with a Text String ........................................................215
E.2 Replacing the Background Image File..........................................................................................215
E.3 Customizing the Product Name....................................................................................................216
E.4 Modifying ini File Parameters via the Web AdminPage ...............................................................217
Appendix F Accessory Programs and Tools..........................................................219
F.1 TrunkPack Downloadable Conversion Utility................................................................................219
F.1.1 Converting a CPT ini File to a Binary dat File.......................................................................220
F.1.2 Encoding / Decoding an ini File ............................................................................................221
F.1.3 Creating a Loadable Prerecorded Tones File.......................................................................222
F.2 Call Progress Tones Wizard .........................................................................................................224
F.2.1 About the Call Progress Tones Wizard.................................................................................224
F.2.2 Installation.............................................................................................................................224
F.2.3 Initial Settings........................................................................................................................224
F.2.4 Recording Screen – Automatic Mode ...................................................................................225
F.2.5 Recording Screen – Manual Mode .......................................................................................226
F.2.6 The Call Progress Tones ini File...........................................................................................227
Appendix G SNMP Traps...........................................................................................229
G.1 Alarm Traps ..................................................................................................................................229
G.1.1 Component: System#0 .........................................................................................................229
G.1.2 Component: AlarmManager#0..............................................................................................231
G.1.3 Component: EthernetLink#0 .................................................................................................231
G.1.4 Other Traps...........................................................................................................................232
G.1.5 Trap Varbinds........................................................................................................................232
Appendix H Regulatory Information ........................................................................233
H.1 MP-11x FXS .................................................................................................................................233
H.2 MP-11x FXO.................................................................................................................................234
H.3 MP-124 .........................................................................................................................................236
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MP-1xx SIP User’s Manual 8 Document #: LTRT-65404
List of Figures
Figure 1-1: MP-124 Gateway Front View .........................................................................................................15
Figure 1-2: MP-108 Gateway Front View .........................................................................................................16
Figure 1-3: MP-104 Gateway Front View .........................................................................................................16
Figure 1-4: MP-102 Gateway Front View .........................................................................................................16
Figure 1-5: Typical MP-1xx VoIP Application ...................................................................................................17
Figure 2-1: MP-108 Front Panel.......................................................................................................................21
Figure 2-2: MP-124 Front Panel.......................................................................................................................21
Figure 2-3: MP-104/FXS Rear Panel Connectors ............................................................................................23
Figure 2-4: MP-124 (FXS) Rear Panel Connectors..........................................................................................23
Figure 3-1: Desktop or Shelf Mounting.............................................................................................................26
Figure 3-2: MP-108 with Brackets for Rack Installation ...................................................................................27
Figure 3-3: MP-124 with Brackets for Rack Installation ...................................................................................28
Figure 3-4: MP-102 Wall Mount........................................................................................................................28
Figure 3-5: RJ-45 Ethernet Connector Pinout ..................................................................................................30
Figure 3-6: RJ-11 Phone Connector Pinout .....................................................................................................30
Figure 3-7: 50-pin Telco Connector (MP-124/FXS only)..................................................................................30
Figure 3-8: MP-124 in a 19-inch Rack with MDF Adaptor................................................................................30
Figure 3-9: DC Power Supply on the MP-124 ..................................................................................................31
Figure 3-10: RS-232 Cable Wiring ...................................................................................................................32
Figure 3-11: Lifeline Splitter Pinout & RJ-11 Connector for MP-10x/FXS........................................................32
Figure 3-12: MP-104/FXS Lifeline Setup..........................................................................................................33
Figure 4-1: Quick Setup Screen .......................................................................................................................37
Figure 5-1: Embedded Web Server Login Screen ...........................................................................................41
Figure 5-2: MP-1xx Web Interface....................................................................................................................42
Figure 5-3: Protocol Definition, General Parameters Screen...........................................................................44
Figure 5-4: Proxy & Registration Parameters Screen ......................................................................................48
Figure 5-5: Coders Screen ...............................................................................................................................53
Figure 5-6: DTMF & Dialing Parameters Screen..............................................................................................55
Figure 5-7: Advanced Parameters, General Parameters Screen ....................................................................58
Figure 5-8: Supplementary Services Parameters Screen................................................................................63
Figure 5-9: Keypad Features Screen ...............................................................................................................66
Figure 5-10: Source Phone Number Manipulation Table for TelIP calls ......................................................68
Figure 5-11: Routing Tables, General Parameters Screen..............................................................................73
Figure 5-12: Tel to IP Routing Table Screen....................................................................................................76
Figure 5-13: IP to Hunt Group Routing Table Screen ......................................................................................78
Figure 5-14: Internal DNS Table Screen ..........................................................................................................80
Figure 5-15: Reasons for Alternative Routing Screen......................................................................................81
Figure 5-16: Coder Group Settings Screen......................................................................................................83
Figure 5-17: Tel Profile Settings Screen...........................................................................................................85
Figure 5-18: IP Profile Settings Screen ............................................................................................................87
Figure 5-19: Endpoint Phone Number Table Screen .......................................................................................89
Figure 5-20: Hunt Group Settings screen.........................................................................................................91
Figure 5-21: Authentication Screen..................................................................................................................93
Figure 5-22: Automatic Dialing Table Screen...................................................................................................94
Figure 5-23: Caller Display Information Screen ...............................................................................................95
Figure 5-24: Call Forwarding Table Screen .....................................................................................................96
Figure 5-25: MP-1xx FXS Caller ID Permission Screen...................................................................................98
Figure 5-26: FXO Settings Screen ...................................................................................................................99
Figure 5-27: Network Settings Screen............................................................................................................103
Figure 5-28: SNMP Managers Table Screen .................................................................................................107
Figure 5-29: Channel Settings, Voice Settings Parameters...........................................................................113
Figure 5-30: Channel Settings, Fax/Modem/CID Parameters........................................................................115
Figure 5-31: Channel Settings, RTP Parameters...........................................................................................117
Figure 5-32: Channel Settings, Miscellaneous Parameters ...........................................................................118
Figure 5-33: Configuration File Screen...........................................................................................................121
Figure 5-34: Regional Settings Screen...........................................................................................................122
Figure 5-35: Change Password Screen .........................................................................................................123
Figure 5-36: IP Connectivity Screen...............................................................................................................124
Figure 5-37: TelIP Call Counters Screen....................................................................................................126
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Figure 5-38: MP-1xx/FXS Channel Status Screen.........................................................................................127
Figure 5-39: Channel Status Details Screen ..................................................................................................128
Figure 5-40: Message Log Screen .................................................................................................................129
Figure 5-41: System Information Screen........................................................................................................130
Figure 5-42: Start Software Upgrade Screen .................................................................................................131
Figure 5-43: Load a cmp File Screen .............................................................................................................132
Figure 5-44: cmp File Successfully Loaded into the MP-1xx Notification ......................................................133
Figure 5-45: Load an ini File Screen ..............................................................................................................134
Figure 5-46: Load a CPT File Screen.............................................................................................................135
Figure 5-47: FINISH Screen ...........................................................................................................................136
Figure 5-48: ‘End Process’ Screen.................................................................................................................136
Figure 5-49: Auxiliary Files Screen.................................................................................................................138
Figure 5-50: Save Configuration Screen ........................................................................................................139
Figure 5-51: Reset Screen .............................................................................................................................140
Figure 6-1: ini File Structure ...........................................................................................................................142
Figure 6-2: SIP ini File Example .....................................................................................................................142
Figure 7-1: Call Progress Tone Types............................................................................................................144
Figure 7-2: Defining a Dial Tone Example .....................................................................................................145
Figure 7-3: Examples of Various Ringing Signals..........................................................................................146
Figure 8-1: Metering Tone Relay Architecture ...............................................................................................159
Figure 8-2: Proprietary Info Message for Relaying Metering Tones ..............................................................159
Figure 8-3: SIP Call Flow................................................................................................................................161
Figure 8-4: MP-108 FXS & FXO Remote IP Extension..................................................................................166
Figure 9-1: Status and Error Messages..........................................................................................................170
Figure 9-2: Setting the Syslog Server IP Address..........................................................................................172
Figure 9-3: The ini File Example for Syslog....................................................................................................172
Figure 10-1: MP-1xx Startup Process ............................................................................................................174
Figure 11-1: Example of Entries in a Device ini file Regarding SNMP...........................................................189
Figure B-1: Main Screen.................................................................................................................................199
Figure B-2: Reset Screen ...............................................................................................................................199
Figure B-3: Preferences Screen .....................................................................................................................201
Figure B-4: Client Configuration Screen.........................................................................................................203
Figure B-5: Templates Screen........................................................................................................................207
Figure E-1: User-Customizable Web Interface Title Bar ................................................................................213
Figure E-2: Customized Web Interface Title Bar............................................................................................213
Figure E-3: Image Download Screen .............................................................................................................214
Figure E-4: INI Parameters Screen ................................................................................................................217
Figure F-1: TrunkPack Downloadable Conversion Utility Opening Screen....................................................219
Figure F-2: Call Progress Tones Conversion Screen.....................................................................................220
Figure F-3: Encode/Decode ini File(s) Screen ...............................................................................................221
Figure F-4: Prerecorded Tones Screen..........................................................................................................222
Figure F-5: File Data Window.........................................................................................................................223
Figure F-6: Initial Settings Screen ..................................................................................................................224
Figure F-7: Recording Screen –Automatic Mode ...........................................................................................225
Figure F-8: Recording Screen after Automatic Detection...............................................................................226
Figure F-9: Recording Screen - Manual Mode ...............................................................................................227
Figure F-10: Call Progress Tone Properties...................................................................................................228
Figure F-11: Call Progress Tone Database Matches.....................................................................................228
Figure F-12: Full PBX/Country Database Match ............................................................................................228
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List of Tables
Table 2-1: Front Panel Buttons on the MP-1xx ................................................................................................21
Table 2-2: Indicator LEDs on the MP-1xx Front Panel.....................................................................................22
Table 2-3: MP-10x Rear Panel Component Descriptions ................................................................................23
Table 2-4: Indicator LEDs on the MP-10x Rear Panel .....................................................................................23
Table 2-5: MP-124 Rear Panel Component Descriptions ................................................................................24
Table 2-6: Indicator LEDs on the MP-124 Rear Panel.....................................................................................24
Table 3-1: Cables and Cabling Procedure .......................................................................................................29
Table 3-2: DC Power Supply on the MP-124 Component Descriptions...........................................................31
Table 3-3: Pin Allocation in the 50-pin Telco Connector ..................................................................................31
Table 3-4: MP-104/FXS Lifeline Setup Component Descriptions ....................................................................33
Table 4-1: MP-1xx Default Networking Parameters .........................................................................................35
Table 5-1: Protocol Definition, General Parameters (continues on pages 45 to 47)........................................45
Table 5-2: Proxy & Registration Parameters (continues on pages 49 to 52) ...................................................49
Table 5-3: ini File Coder Parameter .................................................................................................................54
Table 5-4: DTMF & Dialing Parameters (continues on pages 55 to 57) ..........................................................55
Table 5-5: Advanced Parameters, General Parameters (continues on pages 59 to 62) .................................59
Table 5-6: Supplementary Services Parameters (continues on pages 64 to 65).............................................64
Table 5-7: Keypad Features Parameters .........................................................................................................67
Table 5-8: Number Manipulation Parameters ..................................................................................................69
Table 5-9: Number Manipulation ini File Parameters (continues on pages 70 to 71)......................................70
Table 5-10: Routing Tables, General Parameters (continues on pages 73 to 74)...........................................73
Table 5-11: Tel to IP Routing Table..................................................................................................................76
Table 5-12: IP to Hunt Group Routing Table....................................................................................................79
Table 5-13: Internal DNS ini File Parameter ....................................................................................................80
Table 5-14: Reasons for Alternative Routing ini File Parameter ......................................................................82
Table 5-15: ini File Coder Group Parameters ..................................................................................................84
Table 5-16: ini File Tel Profile Settings.............................................................................................................86
Table 5-17: ini File IP Profile Settings ..............................................................................................................88
Table 5-18: Endpoint Phone Numbers Table ...................................................................................................89
Table 5-19: Channel Select Modes ..................................................................................................................92
Table 5-20: Authentication ini File Parameter ..................................................................................................93
Table 5-21: Automatic Dialing ini File Parameter.............................................................................................95
Table 5-22: Caller ID ini File Parameter ...........................................................................................................96
Table 5-23: Call Forward Table ........................................................................................................................97
Table 5-24: Authentication ini File Parameter ..................................................................................................98
Table 5-25: FXO Parameters (continues on pages 99 to 100) ........................................................................99
Table 5-26: Protocol Management, ini File Parameters (continues on pages 101 to 102) ............................101
Table 5-27: Network Setting Parameters (continues on pages 104 to 106) ..................................................104
Table 5-28: SNMP Managers Table Parameters ...........................................................................................107
Table 5-29: Board, ini File Parameters (continues on pages 108 to 110)......................................................108
Table 5-30: SNMP ini File Parameters...........................................................................................................111
Table 5-31: Channel Settings, Voice Settings Parameters ............................................................................114
Table 5-32: Channel Settings, Fax/Modem/CID Parameters (continues on pages 115 to 116)....................115
Table 5-33: Channel Settings, RTP Parameters ............................................................................................117
Table 5-34: Channel Settings, Miscellaneous Parameters ............................................................................118
Table 5-35: Channel Settings, ini File Parameters.........................................................................................119
Table 5-36: IP Connectivity Parameters.........................................................................................................125
Table 5-37: Call Counters Description (continues on pages 126 to 127).......................................................126
Table 5-38: Auxiliary Files Descriptions .........................................................................................................137
Table 5-39: Configuration Files ini File Parameters .......................................................................................138
Table 8-1: Summary of DTMF configuration Parameters (continues on pages 151 to 152)..........................151
Table 8-2: Supported CDR Fields ..................................................................................................................158
Table 10-1: Vendor Specific Information Field ...............................................................................................176
Table 10-2: Structure of the Vendor Specific Information Field .....................................................................177
Table 12-1: MP-1xx Selected Technical Specifications (continues on pages 191 to 193) ............................191
Table A-1: MP-1xx SIP Supplied Software Kit ...............................................................................................195
Table B-1: Command Line Switch Descriptions .............................................................................................206
Table C-1: Packet Types Defined in RFC 1890 .............................................................................................209
Table C-2: Defined Payload Types.................................................................................................................209
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Table C-3: Default RTP/RTCP/T.38 Port Allocation.......................................................................................210
Table E-1: Customizable Logo ini File Parameters........................................................................................215
Table E-2: Web Appearance Customizable ini File Parameters....................................................................215
Table E-3: Customizable Logo ini File Parameters........................................................................................216
Table E-4: Web Appearance Customizable ini File Parameters....................................................................216
Table G-1: acBoardFatalError Alarm Trap .....................................................................................................229
Table G-2: acBoardEvResettingBoard Alarm Trap ........................................................................................229
Table G-3: acBoardCallResourcesAlarm Alarm Trap ....................................................................................230
Table G-4: acBoardControllerFailureAlarm Alarm Trap .................................................................................230
Table G-5: acBoardOverloadAlarm Alarm Trap .............................................................................................230
Table G-6: acActiveAlarmTableOverflow Alarm Trap ....................................................................................231
Table G-7: acBoardEthernetLinkAlarm Alarm Trap........................................................................................231
Table G-8: coldStart Trap ...............................................................................................................................232
Table G-9: authenticationFailure Trap............................................................................................................232
Table G-10: acBoardEvBoardStarted Trap ....................................................................................................232
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MP-1xx SIP
MP-1xx SIP User’s Manual 12 Document #: LTRT-65404
Tip: When viewing this manual on CD, Web site or on any other electronic copy,
all cross-references are hyperlinked. Click on the page or section numbers (shown in blue) to reach the individual cross-referenced item directly. To return back to the point from where you accessed the cross-reference, press the ALT and keys.
Trademarks
AC logo, Ardito, AudioCoded, AudioCodes, AudioCodes logo, IPmedia, Mediant, MediaPack, MP­MLQ, NetCoder, Stretto, TrunkPack, VoicePacketizer and VoIPerfect, are trademarks or registered trademarks of AudioCodes Limited. All other products or trademarks are property of their respective owners.
Customer Support
Customer technical support and service are provided by AudioCodes’ Distributors, Partners, and Resellers from whom the product was purchased. For Customer support for products purchased directly from AudioCodes, contact support@audiocodes.com
.
Abbreviations and Terminology
Each abbreviation, unless widely used, is spelled out in full when first used. Only industry­standard terms are used throughout this manual. Hexadecimal notation is indicated by 0x preceding the number.
Related Documentation
Document # Manual Name
LTRT-656xx (e.g., LTRT-65601) MP-1xx SIP Release Notes LTRT-614xx MP-1xx Fast Track Installation Guide
Note 1: MP-1xx refers to the MP-124 24-port, MP-108 8-port, MP-104 4-port and
MP-102 2-port media gateways having similar functionality except for the number of channels (the MP-124 and MP-102 support only FXS).
Note 2: MP-10x refers to MP-108 8-port, MP-104 4-port and MP-102 2-port
gateways.
Note 3: MP-1xx/FXS refers only to the MP-124/FXS, MP-108/FXS, MP-104/FXS and
MP-102/FXS gateways.
Note 4: MP-10x/FXO refers only to MP-108/FXO and MP-104/FXO gateways.
Note: Where “network” appears in this manual, it means Local Area Network
(LAN), Wide Area Network (WAN), etc. accessed via the gateway’s Ethernet interface.
Note: FXO (Foreign Exchange Office) is the interface replacing the analog
telephone and connects to a Public Switched Telephone Network (PSTN) line from the Central Office (CO) or to a Private Branch Exchange (PBX). The FXO is designed to receive line voltage and ringing current, supplied from the CO or the PBX (just like an analog telephone). An FXO VoIP gateway interfaces between the CO/PBX line and the Internet.
FXS (Foreign Exchange Station) is the interface replacing the Exchange
(i.e., the CO or the PBX) and connects to analog telephones, dial-up modems, and fax machines. The FXS is designed to supply line voltage and ringing current to these telephone devices. An FXS VoIP gateway interfaces between the analog telephone devices and the Internet.
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MP-1xx SIP User’s Manual General
Version 4.4 13 March 2005
Warning: Ensure that you connect FXS ports to analog telephone or to PBX-trunk
lines only and FXO ports to CO/PBX lines only.
Warning: The MP-1xx is supplied as a sealed unit and must only be serviced by
qualified service personnel.
Warning: Disconnect the MP-1xx from the mains and from the Telephone Network
Voltage (TNV) before servicing.
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MP-1xx SIP User’s Manual 14 Document #: LTRT-65404
Reader’s Notes
Page 15
MP-1xx SIP User’s Manual 1. Overview
Version 4.4 15 March 2005
1 Overview
1.1 Introduction
This document provides you with the information on installation, configuration and operation of the MP-124 24-port, MP-108 8-port, MP-104 4-port and MP-102 2-port VoIP media gateways. As these units have similar functionality, except for the number of channels and some minor features, they are referred to collectively as the MP-1xx. It is expected that the readers are familiar with regular telephony and data networking concepts.
1.2 Gateway Description
The MediaPack MP-1xx Series Analog VoIP gateways are cost-effective, cutting edge technology solutions, providing superior voice quality and optimized packet voice streaming (voice, fax and data traffic) over the same IP network. These gateways use the award-winning, field-proven Digital Signal Processing (DSP) voice compression technology used in other MediaPack and TrunkPack
TM
series products.
The MP-1xx gateways incorporate up to 24 analog ports for connection, either directly to an enterprise PBX (MP-10x/FXO), to phones, or to fax (MP-1xx/FXS), supporting up to 24 simultaneous VoIP calls.
Additionally, the MP-1xx units are equipped with a 10/100 Base-TX Ethernet port for connection to the network.
The MP-1xx gateways are best suited for small to medium size enterprises, branch offices or for residential media gateway solutions.
The MP-1xx gateways enable Users to make free local or international telephone/fax calls between the distributed company offices, using their existing telephones/fax. These calls are routed over the existing network ensuring that voice traffic uses minimum bandwidth.
The MP-1xx gateways are very compact devices that can be installed as a desk-top unit (refer to Section
3.3.1) or on the wall (refer to Section 3.3.4) or in a 19-inch rack (refer to Section 3.3.3
and Section
3.3.3).
The MP-1xx gateways support SIP (Session Initiation Protocol) or H.323 protocols, enabling the deployment of "voice over IP" solutions in environments where each enterprise or residential location is provided with a simple media gateway.
This provides the enterprise with a telephone connection (e.g., RJ-11), and the capability to transmit the voice and telephony signals over a packet network.
The MP-124 supports up to 24 analog telephone loop start FXS ports, shown in Figure
1-1.
Figure
1-1: MP-124 Gateway Front View
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MP-1xx SIP User’s Manual 16 Document #: LTRT-65404
The MP-108 supports up to 8 analog telephone loop start FXS or FXO ports, shown in Figure 1-2.
Figure
1-2: MP-108 Gateway Front View
The MP-104 supports up to 4 analog telephone loop start FXS or FXO ports, shown in Figure
1-3.
Figure
1-3: MP-104 Gateway Front View
The MP-102 supports up to 2 analog telephone loop start FXS ports, shown in Figure
1-4.
Figure
1-4: MP-102 Gateway Front View
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MP-1xx SIP User’s Manual 1. Overview
Version 4.4 17 March 2005
The layout diagram (Figure 1-5), illustrates a typical MP-108 and MP-104 or MP-102 VoIP application.
Figure
1-5: Typical MP-1xx VoIP Application
1.3 SIP Overview
SIP (Session Initialization Protocol) is an application-layer control (signaling) protocol used on the MP-1xx for creating, modifying, and terminating sessions with one or more participants. These sessions can include Internet telephone calls, media announcements and conferences.
SIP invitations are used to create sessions and carry session descriptions that enable participants to agree on a set of compatible media types. SIP uses elements called Proxy servers to help route requests to the user's current location, authenticate and authorize users for services, implement provider call-routing policies and provide features to users.
SIP also provides a registration function that enables users to upload their current locations for use by Proxy servers. SIP, on the MP-1xx, complies with the IETF (Internet Engineering Task Force) RFC 3261 (refer to http://www.ietf.org
).
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1.4 MP-1xx Features
This section provides a high-level overview of some of the many MP-1xx supported features.
1.4.1 General Features
Superior, high quality Voice, Data and fax over IP networks.
Spans a range of 2 to 24 analog ports.
Supports analog telephone sets or analog PSTN/PBX trunk lines (FXS/FXO).
Connects to the network via a 10/100 Base-TX Ethernet interface.
Selectable G.711 or Low Bit Rate (LBR) coders per channel.
T.38 fax with superior performance (handling a round-trip delay of up to nine seconds).
Echo Canceler, Jitter Buffer,
Voice Activity Detection (VAD) and Comfort Noise Generation (CNG)
support.
Web management for easy configuration and installation.
Simple Network Management Protocol (SNMP) and Syslog support.
Simple Network Time Protocol (SNTP) support, the time-of-day can be obtained from a
standard SNTP server.
1.4.2 Hardware Features
MP-124 19-inch, 1 U rugged enclosure provides up to 24 analog FXS ports, using a single
50 pin Telco connector.
MP-10x compact, rugged enclosure only one-half of a 19-inch rack unit, 1 U high (1.75" or
44.5 mm).
MP-124 devices: optional AC or DC power supply.
Lifeline - provides a wired phone connection to PSTN line when there is no power, or the
network fails (applies to MP-10x FXS gateways).
LEDs on the front and rear panels that provide information on the operating status of the
media gateway and the network interface.
Restart button on the Front panel that restarts the MP-1xx gateway, and is also used to
restore the MP-1xx parameters to their factory default values.
1.4.3 SIP Features
The MP-1xx SIP gateway complies with the IETF RFC 3261 standard.
Reliable User Datagram Protocol (UDP) transport, with retransmissions.
T.38 real time fax (using SIP).
Note: If the remote side includes the fax maximum rate parameter in the Session Description Protocol (SDP) body of the Invite message, the gateway returns the same rate in the response SDP.
Works with Proxy or without Proxy, using an internal routing table.
Fallback to internal routing table if Proxy is not responding.
Supports up to four Proxy servers. If the primary Proxy fails, the MP-1xx automatically
switches to a redundant Proxy.
Supports Proxy server discovery using Domain Name Server (DNS) SRV records.
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MP-1xx SIP User’s Manual 1. Overview
Version 4.4 19 March 2005
Proxy and Registrar Authentication (handling 401 and 407 responses) using Basic or Digest
methods.
Single gateway Registration or multiple Registration of all gateway endpoints.
Configuration of authentication username and password per each gateway endpoint, or
single username and password per gateway.
Supported methods: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, INFO, REFER,
NOTIFY, PRACK, UPDATE and SUBSCRIBE.
Modifying connection parameters in a call (re-INVITE).
Working with Redirect server and handling 3xx responses.
Early media (supporting 183 Session Progress).
PRACK reliable provisional responses <RFC 3262>.
Call Hold and Transfer Supplementary services using REFER, Refer-To, Referred-By,
Replaces and NOTIFY.
Call Forward (using 302 response): Immediate, Busy, No reply, Busy or No reply, Do Not
Disturb.
Supports RFC 3327 – Adding “Path” to Supported header.
Supports RFC 3581 – Symmetric Response Routing.
Session Timer <draft-ietf-sip-session-timer-13.txt>.
Supports network asserted identity and privacy (RFC 3325 and RFC 3323).
Supports Tel URI (Uniform Resource Identifier) according to RFC 2806 bis.
Remote party ID <draft-ietf-sip-privacy-04.txt>.
Obtaining Proxy Domain Name(s) from DHCP (Dynamic Host Control Protocol) according to
RFC 3361.
RFC 2833 relay for Dual Tone Multi Frequency (DTMF) digits, including payload type
negotiation.
DTMF out-of-band transfer using:
INFO method <draft-choudhuri-sip-info-digit-00.txt>. INFO method, compatible with Cisco gateways. NOTIFY method <draft-mahy-sipping-signaled-digits-01.txt>.
Supported coders:
G.711 A-law 64 kbps (10, 20, 30, 40, 50, 60, 80, 100, 120 msec)
G.711 µ-law 64 kbps (10, 20, 30, 40, 50, 60, 80, 100, 120 msec) G.723.1 5.3, 6.3 kbps (30, 60, 90, 120, 150 msec)
G.726 32 kbps (10, 20, 30, 40, 50, 60, 80, 100, 120 msec)
G.729A 8 kbps (10, 20, 30, 40, 50, 60, 80, 100, 120 msec)
G.729B is supported if Silence Suppression is enabled.
Can negotiate coder from a list of given coders.
Implementation of Message Waiting Indication (MWI) IETF draft-ietf-sipping-mwi-04.txt,
including SUBSCRIBE (to the MWI server). The MP-1xx/FXS gateways can accept an MWI Notify message that indicates waiting messages or indicates that the MWI is cleared.
For more updated information on the gateway’s supported features, refer to the latest MP-1xx SIP Release Notes.
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Reader’s Notes
Page 21
MP-1xx SIP User’s Manual 2. MP-1xx Physical Description
Version 4.4 21 March 2005
2 MP-1xx Physical Description
This section provides detailed information on the MP-1xx hardware, the location and functionality of the LEDs, buttons and connectors on the front and rear panels.
For detailed information on installing the MP-1xx refer to Section
3 on page 25.
2.1 MP-1xx Front Panel
Figure 2-1 and Figure 2-2 illustrate the front layout of the MP-108 (almost identical on MP-104
and MP-102) and MP-124 respectively. Refer to Section
2.1.1 for meaning of the front panel
buttons; refer to Section
2.1.2 for functionality of the front panel LEDs.
Figure
2-1: MP-108 Front Panel
Figure
2-2: MP-124 Front Panel
2.1.1 MP-1xx Front Panel Buttons
Table 2-1 lists and describes the front panel buttons on the MP-1xx.
Table
2-1: Front Panel Buttons on the MP-1xx
Type Function Comment
Reset the MP-1xx
Press the reset button with a paper clip or any other similar pointed object, until the gateway is reset.
Reset button
Restore the MP-1xx parameters to
their factory default values
Refer to Section
4.2 on page 36.
Reset Button
Reset Button
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MP-1xx SIP User’s Manual 22 Document #: LTRT-65404
2.1.2 MP-1xx Front Panel LEDs
Table 2-2 lists and describes the front panel LEDs on the MP-1xx.
Note: MP-1xx (FXS/FXO) media gateways feature almost identical front panel
LEDs; they only differ in the number of channel LEDs that correspond to the number of channels.
Table 2-2: Indicator LEDs on the MP-1xx Front Panel
Label Type Color State Function
Green ON
Device Powered, self-test OK
Orange Blinking
Software Loading/Initialization
Ready
Device Status
Red ON
Malfunction
Green ON
Valid 10/100 Base-TX Ethernet connection
LAN
Ethernet Link
Status
Red ON
Malfunction
Green Blinking
Sending and receiving SIP messages
Control
Control Link
Blank
No traffic
Green Blinking
Transmitting RTP (Real-Time Transport Protocol) Packets
Red Blinking
Receiving RTP Packets
Data
Packet Status
Blank -
No traffic Offhook / Ringing for FXS Phone Port
Green ON
FXO Line-Seize/Ringing State for Line Port
Green Blinking
There’s an incoming call, before answering
Red ON
Line Malfunction
Channels
Telephone
Interface
Blank -
Normal
2.2 MP-1xx Rear Panel
2.2.1 MP-10x Rear Panel
Figure 2-3 illustrates the rear panel layout of the MP-104. For descriptions of the MP-10x rear
panel components, refer to Table
2-3. For the functionality of the MP-10x rear panel LEDs, refer
to Table
2-4.
Tip 1: MP-10x (FXS/FXO) media gateways feature almost identical rear panel
connectors and LEDs, located slightly differently from one device to the next.
Tip 2: The RJ-45 port (Eth 1) on the MP-10x/FXO rear panel is inverted on the MP-
1xx/FXS. The label on the rear panel also distinguishes FXS from FXO devices.
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MP-1xx SIP User’s Manual 2. MP-1xx Physical Description
Version 4.4 23 March 2005
Figure 2-3: MP-104/FXS Rear Panel Connectors
Table 2-3: MP-10x Rear Panel Component Descriptions
Item # Label Component Description
1
100-250V ~ 1A
50-60 Hz
AC power supply socket.
2
Protective earthing screw (mandatory for all installations).
3 Eth 1
10/100 Base-TX Ethernet connection.
4
2, 4 or 8 FXS/FXO ports.
5 FXS
FXS / FXO label.
6 RS-232
9 pin RS-232 status port (for Cable Wiring of the RS-232 refer to Figure
3-10 on page 32).
Table 2-4: Indicator LEDs on the MP-10x Rear Panel
Label Type Color State Meaning
Yellow ON
Ethernet port receiving data
ETH-1
Ethernet Status
Red ON
Collision
Note that the Ethernet LEDs are located within the RJ-45 socket.
2.2.2 MP-124 Rear Panel
Figure 2-4 illustrates the rear panel layout of the MP-124. For descriptions of the MP-124 rear
panel components, refer to Table
2-5. For the functionality of the MP-124 rear panel LEDs, refer
to Table
2-6.
Figure
2-4: MP-124 (FXS) Rear Panel Connectors
1
2 3
4
6
5
1
2
3
4
6
5
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Table 2-5: MP-124 Rear Panel Component Descriptions
Item # Label Component Description
1
Protective earthing screw (mandatory for all installations).
2
100-250 V~
50 - 60 Hz 2A
AC power supply socket.
3 ANALOG LINES 1 –24
50-pin Telco for 1 to 24 analog lines.
4 Data Cntrl Ready
LED indicators (described in Table 2-6).
5 RS-232
9 pin RS-232 status port (for Cable Wiring of the RS-232 refer to Figure
3-10 on page 32).
6 Eth 1 Eth 2
Dual 10/100 Base-TX Ethernet connections.
48 V
2A max
Connection to external DC 40-60 V power supply (refer to Figure
3-9).
Note: The Dual In-line Package (DIP) switch, located on the MP-124 rear panel
(supplied with some of the units), is not functional and should not be used.
The Ethernet LEDs are located within each of the RJ-45 sockets.
Note that on the MP-124 the rear panel also duplicates the Data, Control and Ready LEDs from the front panel.
Table
2-6: Indicator LEDs on the MP-124 Rear Panel
Label Type Color State Function
Green ON
Transmitting RTP Packets
Red ON
Receiving RTP Packets
Data
Packet Status
Blank
No traffic
Green Blinking
Sending and receiving H.323 messages
Cntrl
Control Link
Blank
No traffic
Green ON
Device Powered and Self-test OK
Orange ON
Software Loading/Initialization
Ready
Device Status
Red ON
Malfunction
Green ON
Valid 10/100 Base-TX Ethernet connection
Eth 1
Ethernet Status
Red ON
Malfunction
Green ON
Valid 10/100 Base-TX Ethernet connection
Eth 2
Ethernet Status
Red ON
Malfunction
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MP-1xx SIP User’s Manual 3. Installing the MP-1xx
Version 4.4 25 March 2005
3 Installing the MP-1xx
This section provides information on the hardware installation procedure for the MP-1xx. For information on how to start using the gateway, refer to Section
4 on page 35. For detailed
information on the MP-1xx connectors, LEDs and buttons, refer to Section
2 on page 21.
Caution Electrical Shock
The equipment must only be installed or serviced by qualified service personnel.
To install the MP-1xx, take these 4 steps:
1. Unpack the MP-1xx (refer to Section 3.1 below).
2. Check the package contents (refer to Section
3.2 below).
3. Mount the MP-1xx (refer to Section
3.3 on page 26).
4. Cable the MP-1xx (refer to Section
3.4 on page 29).
After connecting the MP-1xx to the power source, the Ready and LAN LEDs on the front panel turn to green (after a self-testing period of about 1 minute). Any malfunction changes the Ready LED to red.
When you have completed the above relevant sections you are then ready to start configuring the gateway (Section
4 on page 35).
3.1 Unpacking
To unpack the MP-1xx, take these 6 steps:
1. Open the carton and remove packing materials.
2. Remove the MP-1xx gateway from the carton.
3. Check that there is no equipment damage.
4. Check, retain and process any documents.
5. Notify AudioCodes or your local supplier of any damage or discrepancies.
6. Retain any diskettes or CDs.
3.2 Package Contents
Ensure that in addition to the MP-1xx, the package contains:
AC power cable for the AC power supply option.
DC terminal block (MP-124 only, for the DC power supply option).
CD (software and documentation).
Lifeline cable (RJ-11 adaptor cable for 1 to 2). (Supplied with MP-10x/FXS only).
3 brackets (2 short, 1 long) and bracket-to-device screws for 19-inch rack installation option
(MP-10x only).
2 short equal-length brackets and bracket-to-device screws for MP-124 19-inch rack
installation.
The MP-1xx Fast Track Installation Guide.
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3.3 Mounting the MP-1xx
The MP-1xx can be mounted on a desktop or on a wall (only MP-10x), or installed in a standard 19-inch rack. Refer to Section
3.4 on page 29 for cabling the MP-1xx.
3.3.1 Mounting the MP-1xx on a Desktop
No brackets are required. Simply place the MP-1xx on the desktop in the position you require.
Figure
3-1: Desktop or Shelf Mounting
Rack Mount Safety Instructions (UL)
When installing the chassis in a rack, be sure to implement the following Safety instructions recommended by Underwriters Laboratories:
Elevated Operating Ambient - If installed in a closed or multi-unit rack assembly,
the operating ambient temperature of the rack environment may be greater than room ambient. Therefore, consideration should be given to installing the equipment in an environment compatible with the maximum ambient temperature (Tma) specified by the manufacturer.
Reduced Air Flow - Installation of the equipment in a rack should be such that the
amount of air flow required for safe operation on the equipment is not compromised.
Mechanical Loading - Mounting of the equipment in the rack should be such that
a hazardous condition is not achieved due to uneven mechanical loading.
Circuit Overloading - Consideration should be given to the connection of the
equipment to the supply circuit and the effect that overloading of the circuits might have on overcurrent protection and supply wiring. Appropriate consideration of equipment nameplate ratings should be used when addressing this concern.
Reliable Earthing - Reliable earthing of rack-mounted equipment should be
maintained. Particular attention should be given to supply connections other than direct connections to the branch circuit (e.g., use of power strips.)
3.3.2 Installing the MP-10x in a 19-inch Rack
The MP-10x is installed into a standard 19-inch rack by the addition of two supplied brackets (1 short, 1 long). The MP-108 with brackets for rack installation is shown in Figure
3-2.
To install the MP-10x in a 19-inch rack, take these 9 steps:
1. Remove the two screws on one side of the device nearest the front panel.
2. Insert the peg on the short bracket into the third air vent down on the column of air vents
nearest the front panel.
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MP-1xx SIP User’s Manual 3. Installing the MP-1xx
Version 4.4 27 March 2005
3. Swivel the bracket until the holes in the bracket line up with the two empty screw holes on
the device.
4. Use the screws found in the devices’ package to attach the short bracket to the side of the
device.
5. Remove the two screws on the other side of the device nearest the front panel.
6. Position the long bracket so that the holes in the bracket line up with the two empty screw
holes on the device.
7. Use the screws found in the device’s package to attach the long bracket to the side
of the
device.
8. Position the device in the rack and line up the bracket holes with the rack frame holes.
9. Use four standard rack screws to attach the device to the rack. These screws are not
provided with the device.
Figure
3-2: MP-108 with Brackets for Rack Installation
3.3.3 Installing the MP-124 in a 19-inch Rack
The MP-124 is installed into a standard 19-inch rack by the addition of two short (equal-length) supplied brackets. The MP-124 with brackets for rack installation is shown in Figure
3-3.
To install the MP-124 in a 19-inch rack, take these 7 steps:
1. Remove the two screws on one side of the device nearest the front panel.
2. Insert the peg on one of the brackets into the third air vent down on the column of air vents
nearest the front panel.
3. Swivel the bracket until the holes in the bracket line up with the two empty screw holes on
the device.
4. Use the screws found in the devices’ package to attach the bracket to the side of the device.
5. Repeat steps 1 to 4 to attach the second bracket to the other side of the device.
6. Position the device in the rack and line up the bracket holes with the rack frame holes.
7. Use four standard rack screws to attach the device to the rack. These screws are not
provided with the device.
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MP-1xx SIP
MP-1xx SIP User’s Manual 28 Document #: LTRT-65404
Figure 3-3: MP-124 with Brackets for Rack Installation
3.3.4 Mounting the MP-10x on a Wall
The MP-10x is mounted on a wall by the addition of two short (equal-length) supplied brackets. The MP-102 with brackets for wall mount is shown in Figure
3-4.
To mount the MP-10x on a wall, take these 7 steps:
1. Remove the screw on the side of the device that is nearest the bottom and the front panel.
2. Insert the peg on the bracket into the third air vent down on the column of air vents nearest
the front panel.
3. Swivel the bracket so that the side of the bracket is aligned with the base of the device and
the hole in the bracket line up with the empty screw hole.
4. Attach the bracket using one of the screws provided in the device package.
5. Repeat steps 1 to 4 to attach the second bracket to the other side of the device.
6. Position the device on the wall with the base of the device next to the wall.
7. Use four screws to attach the device to the wall. These screws are not provided with the
device.
Figure
3-4: MP-102 Wall Mount
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MP-1xx SIP User’s Manual 3. Installing the MP-1xx
Version 4.4 29 March 2005
3.4 Cabling the MP-1xx
Verify that you have the cables listed under column ‘Cable’ in Table 3-1 before beginning to cable the MP-1xx according to the column ‘Cabling Procedure’. For detailed information on the MP-1xx
rear panel connectors, refer to Section
2.2 on page 22.
Table
3-1: Cables and Cabling Procedure
Cable Cabling Procedure
RJ-45 Ethernet cable
When initializing (connecting the MP-1xx to the network for the first time) use a standard Ethernet cable to connect the network interface on your computer to a port on a network hub / switch. Use a second standard Ethernet cable to connect the MP-1xx to another port on the same network hub / switch. For normal use, connect the MP-1xx Ethernet connection directly to the network, using a standard RJ-45 Ethernet cable. For connector’s pinout refer to Figure 3-5 on page 30.
Connect the RJ-11 connectors on the rear panel of the MP­10x/FXS to fax machine, modem, or phones (refer to Figure 3-6).
Connect RJ-11 connectors on the MP-10x/FXO rear panel to telephone exchange analog lines or PBX extensions (Figure 3-6).
Ensure that FXS & FXO are connected to the correct devices, otherwise damage can occur.
RJ-11 two­wire telephone cords
MP-124/FXS ports are usually distributed using an MDF Adaptor Block (special order option). Refer to Figure 3-8 for details.
Lifeline cable
For detailed information on setting up the Lifeline, refer to the procedure under Section
3.4.2 on page 32.
50-pin Telco cable (MP-124 devices only).
An Octopus cable is not included with the MP-124 package.
1. Wire the 50-pin Telco connectors according to the pinout in Figure 3-7 on page 30,
and Figure 3-8 on page 30.
2. Attach each pair of wires from a 25-pair Octopus cable to its corresponding socket
on the MDF Adaptor Block’s rear.
3. Connect the wire-pairs at the other end of the cable to a male 50-pin Telco
connector.
4. Insert and fasten this connector to the female 50-pin Telco connector on the MP-
124 rear panel (labeled Analog Lines 1-24).
5. Connect the telephone lines from the Adaptor Block to a fax machine, modem, or
telephones by inserting each RJ-11 connector on the 2-wire line cords of the POTS phones into the RJ-11 sockets on the front of an MDF Adaptor Block as shown in
Figure 3-8 on page 30.
RS-232 serial cable
For detailed information on connecting the MP-1xx RS-232 port to your PC, refer to Section 3.4.1 on page 32.
Protective earthing strap
Connect an earthed strap to the chassis protective earthing screw and fasten it securely according to the safety standards.
AC Power cable
Connect the MP-1xx 100-250V~ 50-60 Hz power socket to the mains.
DC Power cable (MP-124 devices only)
Refer to Figure 3-9. Insert two 18 AWG wires into the supplied DC terminal block and fasten the two screws located directly above each wire. Insert the DC terminal block into the DC inlet on the MP-124 rear panel and fasten it with the two adaptor-to-panel screws. Connect the other end of the cable to a 48 VDC power supply.
Safety Notice
When installing the DC power supply on the MP-124, be sure to implement the following safety instructions:
Connect the unit to an SELV source sufficiently isolated from the mains.
Connect the unit permanently to earth via its protective earthing stud.
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Figure 3-5: RJ-45 Ethernet Connector Pinout
1 2 3 4 5 6 7 8
RJ-45 Connector and Pinout
4, 5, 7, 8
not
connected
1 - Tx+ 2 - Tx­3 - Rx+ 6 - Rx-
Figure
3-6: RJ-11 Phone Connector Pinout
1 2 3 4
1 ­2 ­3 ­4 -
Not connected
RJ-11 Connector and Pinout
Not connected
Tip Ring
Figure
3-7: 50-pin Telco Connector (MP-124/FXS only)
125
26
50
Pin Numbers
Figure
3-8: MP-124 in a 19-inch Rack with MDF Adaptor
Data Cntrl Ready
CONFIG
1 2 3 4 5
ON
RS-232
Eth 1 Eth 2
ANALOG LINES 1- 20
100 - 250V~ 50 - 60Hz 2A
AC Power Cord
Grounding Strap
RS-232 Cable
50-pin female
Telco connector
Back-up
LAN Cable
to Eth 2
Connect to
here
MP-124
Rear View
REAR OUTPUT
24 wire pairs in
Octopus cable with 50-pin male Telco connector
FRONT INPUT
24 line cords
2-wire with RJ-11
connectors
M D F Ad aptor B lock - rear
Primary
LAN Cable
to Eth 1
19-inch Rack
Rear View
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Figure 3-9: DC Power Supply on the MP-124
Table
3-2: DC Power Supply on the MP-124 Component Descriptions
Item #
Component Description
1 2 screws for wire connection to the DC terminal block.
2 2 screws for connecting the DC terminal block to the MP-124 panel.
3 Two 18 AWG wires.
Table 3-3: Pin Allocation in the 50-pin Telco Connector
Phone Channel Connector Pins Phone Channel Connector Pins
1 1/26 13 13/38 2 2/27 14 14/39 3 3/28 15 15/40 4 4/29 16 16/41 5 5/30 17 17/42 6 6/31 18 18/43 7 7/32 19 19/44 8 8/33 20 20/45
9 9/34 21 21/46 10 10/35 22 22/47 11 11/36 23 23/48 12 12/37 24 24/49
1
2
3
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3.4.1 Connecting the MP-1xx RS-232 Port to Your PC
Using a standard RS-232 straight cable (not a cross-over cable) with DB-9 connectors, connect the MP-1xx RS-232 port to either COM1 or COM2 RS-232 communication port on your PC. The
required connector pinout and gender are shown below in Figure
3-10.
The RS-232 port is mainly used internally by service personnel for monitoring purposes. Advanced users can also use this feature to obtain log information (for example).
Figure 3-10: RS-232 Cable Wiring
2 3 5
2 3 5
RD
TD
GND
DB-9 female for PC DB-9 male for MP-100
3.4.1.1 Configuring the Serial Connection
To communicate with the MP-1xx, set your serial communication software to the following communications port settings:
Baud Rate: 115,200 bps
Data bits: 8
Parity: None
Stop bits: 1
Flow control: Hardware
3.4.2 Cabling the Lifeline Phone
The Lifeline provides a wired analog POTS phone connection to any PSTN or PBX FXS port when there is no power, or when the network connection fails. Users can therefore use the Lifeline phone even when the MP-1xx is not powered on or not connected to the network. With the MP-108/FXS and MP-104/FXS the Lifeline connection is provided on port #4 (refer to Figure
3-12). With the MP-102/FXS the Lifeline connection is provided on port #2.
Note: The MP-124 and MP-10x/FXO do NOT support the Lifeline.
The Lifeline’s Splitter connects pins #1 and #4 to another source of an FXS port, and pins #2 and #3 to the POTS phone. Refer to the Lifeline Splitter pinout in Figure
3-11.
Figure
3-11: Lifeline Splitter Pinout & RJ-11 Connector for MP-10x/FXS
1 2 3 4
1 ­2 ­3 ­4 -
Life Line Tip
Life Line Ring
Tip Ring
DB-9 female for PC
DB-9 male for MP-1xx
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To cable the MP-10x/FXS Lifeline phone, take these 3 steps:
1. Connect the Lifeline Splitter to port #4 (on the MP-104/FXS or MP-108/FXS) or to port #2 (on
the MP-102/FXS).
2. Connect the Lifeline phone to Port A on the Lifeline Splitter.
3. Connect an analog PSTN line to Port B on the Lifeline Splitter.
Note: The use of the Lifeline on network failure can be disabled using the
‘LifeLineType’ ini file parameter (described in Table 5-29 on page 108).
Figure 3-12: MP-104/FXS Lifeline Setup
Table
3-4: MP-104/FXS Lifeline Setup Component Descriptions
Item # Component Description
1
B: To PSTN wall port.
2
Phone to Port 1.
3
Lifeline to Port 4.
4
PSTN to Splitter (B).
5
Phone to Port 1.
6
Lifeline phone to Splitter (A).
7
Lifeline phone.
1
2
3
4
5
6
7
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Reader’s Notes
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Version 4.4 35 March 2005
4 Getting Started
The MP-1xx is supplied with application software already resident in its flash memory (with factory default parameters).
Section
4.1 below describes how to assign an IP address to the MP-1xx, while Section 4.2 on
page 36 describes how to set up the MP-1xx with basic parameters using a standard Web browser (such as Microsoft
TM
Internet Explorer).
For detailed information on how to fully configure the gateway refer to the Web Interface, described in Section
5 on page 39.
4.1 Assigning the MP-1xx IP Address
To assign an IP address to the MP-1xx use one of the following methods:
HTTP using a Web browser (refer to Section
4.1.1 below).
BootP (refer to Section
4.1.2 on page 36).
DHCP (refer to Section
10.2 on page 175).
Serial communication software (e.g., HyperTerminal
TM
) connected to the MP-1xx via the RS-
232 port (refer to Section 9.2.2 on page 170).
The default networking parameters are show in Table
4-1.
You can use the ‘Reset’ button to restore the MP-1xx networking parameters to their factory default values (refer to Section
4.2 on page 36).
Table
4-1: MP-1xx Default Networking Parameters
FXS or FXO Default Value
FXS
10.1.10.10
FXO
10.1.10.11
MP-1xx default subnet mask is 255.255.0.0, default gateway IP address is 0.0.0.0
4.1.1 Assigning an IP Address Using HTTP
To assign an IP address using HTTP, take these 8 steps:
1. Connect your computer to the MP-1xx. Either connect the network interface on your
computer to a port on a network hub / switch (refer to Table
3-1 on page 29 - RJ-45 Ethernet
cable), or use an Ethernet cross-over cable to directly connect the network interface on your computer to the RJ-45 jack on the MP-1xx.
2. Change your PC’s IP address and subnet mask to correspond with the MP-1xx factory
default IP address and subnet mask, shown in Table
4-1. For details on changing the IP
address and subnet mask of your PC, refer to Windows™ Online Help (Start>Help).
3. Access the MP-1xx Embedded Web Server (refer to Section
5.5 on page 40).
4. In the ‘Quick Setup’ screen (shown in Figure
4-1), set the MP-1xx ‘IP Address’, ‘Subnet
Mask’ and ‘Default Gateway IP Address’ fields under ‘IP Configuration’ to correspond with your network IP settings. If your network doesn’t feature a default gateway, enter a dummy value in the ‘Default Gateway IP Address’ field.
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5. Click the Reset button and click OK in the prompt; the MP-1xx applies the changes and
restarts. This takes approximately 1 minute to complete. When the MP-1xx has finished restarting, the Ready and LAN LEDs on the front panel are lit green.
Tip: Record and retain the IP address and subnet mask you assign the MP-1xx.
Do the same when defining new username or password. If the Embedded Web Server is unavailable (for example, if you’ve lost your username and password), use the BootP/TFTP (Trivial File Transfer Protocol) configuration utility to access the device, “reflash” the load and reset the password (refer
to Appendix B on page 197 for detailed information on using a BootP/TFTP configuration utility to access the device).
6. Disconnect your computer from the MP-1xx or from the hub / switch (depending on the
connection method you used in step
1).
7. Reconnect the MP-1xx and your PC (if necessary) to the LAN.
8. Restore your PC’s IP address & subnet mask to what they originally were. If necessary,
restart your PC and re-access the MP-1xx via the Embedded Web Server with its new assigned IP address.
4.1.2 Assigning an IP Address Using BootP
Note: BootP procedure can also be performed using any standard compatible
BootP server.
Tip: You can also use BootP to load the auxiliary files to the MP-1xx (refer to
Section 5.11.2.1 on page 138).
To assign an IP address using BootP, take these 3 steps:
1. Open the BootP application (supplied with the MP-1xx software package).
2. Add client configuration for the MP-1xx, refer to Section
B.11.1 on page 203.
3. Reset the gateway physically causing it to use BootP; the MP-1xx changes its network
parameters to the values provided by the BootP.
4.2 Restoring Networking Parameters to their Initial State
You can use the ‘Reset’ button to restore the MP-1xx networking parameters to their factory default values (described in Table
4-1) and to reset the username and password.
Note that the MP-1xx returns to the software version burned in flash. This process also restores the MP-1xx parameters to their factory settings, therefore you must load your previously backed­up ini file, or the default ini file (received with the software kit) to set them to their correct values.
To restore networking parameters to their initial state, take these 6
steps:
1. Disconnect the MP-1xx from the power and network cables.
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2. Reconnect the power cable; the gateway is powered up. After approximately 45 seconds the
Ready LED turns to green and the Control LED blinks for about 3 seconds.
3. While the Control LED is blinking, press shortly on the reset button (located on the left side
of the front panel); the gateway resets a second time and is restored with factory default parameters (username: “Admin”, password: “Admin”).
4. Reconnect the network cable.
5. Assigning the MP-1xx IP address (refer to Section
4.1 on page 35).
6. Load your previously backed-up ini file, or the default ini file (received with the software kit).
To load the ini file via the Embedded Web Server, refer to Section
5.9.2.1 on page 120.
4.3 Configure the MP-1xx Basic Parameters
To configure the MP-1xx basic parameters use the Embedded Web Server’s ‘Quick Setup’ screen (shown in Figure
4-1 below). Refer to Section 5.5 on page 40 for information on accessing
the ‘Quick Setup’ screen.
Figure
4-1: Quick Setup Screen
To configure basic SIP parameters, take these 9 steps:
1. If the MP-1xx is behind a router with Network Address Translation (NAT) enabled, perform
the following procedure. If it isn’t, leave the ‘NAT IP Address’ field undefined.
Determine the “public” IP address assigned to the router (by using, for instance, router
Web management). Enter this public IP address in the ‘NAT IP Address’ field.
Enable the DMZ (Demilitarized Zone) configuration on the residential router for the LAN
port where the MP-1xx gateway is connected. This enables unknown packets to be routed to the DMZ port.
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2. Under ‘SIP Parameters’, enter the MP-1xx Domain Name in the field ‘Gateway Name’. If the
field is not specified, the MP-1xx IP address is used instead (default).
3. When working with a Proxy server, set ‘Working with Proxy’ field to ‘Yes’ and enter the IP
address of the primary Proxy server in the field ‘Proxy IP Address’. When no Proxy is used, the internal routing table is used to route the calls.
4. Enter the Proxy Name in the field ‘Proxy Name’. If Proxy name is used, it replaces the Proxy
IP address in all SIP messages. This means that messages are still sent to the physical Proxy IP address but the SIP URI contains the Proxy name instead.
5. Configure ‘Enable Registration’ to ‘Yes’ or ‘No’:
‘No’ = the MP-1xx does not register to a Proxy server/Registrar (default). ‘Yes’ = the MP-1xx registers to a Proxy server/Registrar at power up and every ‘Registration Time’ seconds; The MP-1xx sends a register request according to the ‘Authentication Mode’ parameter. For detailed information on the parameters ‘Registration Time’ and
‘Authentication Mode’, refer to Table
5-2 on page 49.
6. Select the coder (i.e., vocoder) that best suits your VoIP system requirements. The default
coder is: G.7231 30 msec. To program the entire list of coders you want the MP-1xx to use, click the button on the left side of the ‘1
st
Coder’ field; the drop-down list for the 2nd to 5th coders appears. Select coders according to your system requirements. Note that coders higher on the list are preferred and take precedence over coders lower on the list.
Note: The preferred coder is the coder that the MP-1xx uses as a first choice for all
connections. If the far end gateway does not use this coder, the MP-1xx negotiates with the far end gateway to select a coder that both sides can use.
7. To program the Tel to IP Routing Table, press the arrow button next to ‘Tel to IP Routing
Table’. For information on how to configure the Tel to IP Routing Table, refer to Section
5.8.4.2 on page 75.
8. To program the Endpoint Phone Number Table, press the arrow button next to ‘Endpoint
Phone Number’. For information on how to configure the Endpoint Phone Number Table, refer to Section
5.8.6 on page 89.
9. Click the Reset button and click OK in the prompt; The MP-1xx applies the changes and
restarts. This takes approximately 1 minute to complete. When the MP-1xx has finished restarting, the Ready and LAN LEDs on the front panel are lit green.
You are now ready to start using the VoIP gateway. To prevent unauthorized access to the MP- 1xx, it is recommended that you change the username and password that are used to access the
Web Interface. Refer to Section
5.9.5 on page 123 for details on how to change the username
and password.
Tip: Once the gateway is configured correctly back up your settings by making a
copy of the VoIP gateway configuration (ini file) and store it in a directory on your computer. This saved file can be used to restore configuration settings at a future time. For information on backing up and restoring the gateway’s
configuration refer to Section 5.9.2.1 on page 120.
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5 Configuring the MP-1xx
5.1 Configuration Concepts
Users can utilize the the MP-1xx in a wide variety of applications, enabled by its parameters and configuration files (e.g., Call Progress Tones (CPT), etc.). The parameters can be configured and configuration files can be loaded using:
A standard Web Browser (described and explained in this section).
A configuration file referred to as the ini file. For information on how to use the ini file refer to
Section
6 on page 141.
An SNMP browser software (refer to Section
11 on page 179).
AudioCodes’ Element Management System (EMS) (refer to Section
11.8 on page 190 and to
AudioCodes’ EMS User’s Manual or EMS Product Description).
To upgrade the MP-1xx (load new software or configuration files onto the gateway) use the Software Upgrade wizard, available through the Web Interface (refer to Section
5.11.1 on page
131), or alternatively use the BootP/TFTP configuration utility (refer to Section
10.3.1 on page
176).
For information on the configuration files refer to Section
6 on page 141.
5.2 Overview of the Embedded Web Server
The Embedded Web Server is used both for gateway configuration, including loading of configuration files, and for run-time monitoring. The Embedded Web Server can be accessed from a standard Web browser, such as Microsoft™ Internet Explorer, Netscape™ Navigator, etc. Specifically, Users can employ this facility to set up the gateway configuration parameters. Users also have the option to remotely reset the gateway and to permanently apply the new set of parameters.
5.3 Computer Requirements
To use the Web Interface, the following is needed:
A computer capable of running your Web browser.
A network connection to the VoIP gateway.
One of the following compatible Web browsers:
Microsoft™ Internet Explorer™ (version 6.0 and higher). Netscape™ Navigator™ (version 7.0 and higher).
Note: Some Java-script applications are not supported in Netscape.
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5.4 Password Control
The Embedded Web Server is protected by a unique username and password combination. The first time a browser request is made, the User is requested to provide his username and password to obtain access. Subsequent requests are negotiated by the browser on behalf of the User, so that the User doesn’t have to re-enter the username and password for each request, but the request is still authenticated (the Embedded Web Server uses the MD5 authentication method supported by the HTTP 1.1 protocol).
An additional level of protection is obtained by a restriction that no more than three IP addresses can access the Embedded Web Server concurrently. With this approach, a fourth User is told that the server is busy, even if the correct username and password were provided.
5.4.1 Embedded Web Server Username & Password
The default username and password for all gateways are:
Username = “Admin” (case-sensitive)
Password = “Admin” (case-sensitive)
For details on changing the username and password, refer to Section
5.9.5 on page 123. Note
that the password and username can be a maximum of 7 case-sensitive characters. The User can reset the Web username and password (to the default values) by enabling an ini
file parameter called ‘ResetWebPassword’. The Web password is automatically the default password.
5.5 Configuring the Web Interface via the ini File
Two additional security preferences can be configured using ini file parameters. These security levels provide protection against unauthorized access (such as Internet hacker attacks),
particularly to Users without a firewall. For information on the ini file refer to Section
6 on page
141.
5.5.1 Limiting the Embedded Web Server to Read-Only Mode
Users can limit the Web Interface to read-only mode by changing the ini file parameter ‘DisableWebConfig’ to 1. In this mode all Web screens are read-only and cannot be modified. In addition, the following screens cannot be accessed: ‘Quick Setup’, ‘Change Password’, ’Reset‘, ‘Save Configuration‘, ‘Software Upgrade Wizard’, ‘Load Auxiliary Files’, ‘Configuration File’ and ‘Regional Settings’.
5.5.2 Disabling the Embedded Web Server
To deny access to the gateway through HTTP protocol, the User can disable the Embedded Web Server task. To disable the Web task, use the ini file parameter ‘DisableWebTask = 1’. The default is to Web task enabled.
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5.6 Accessing the Embedded Web Server
To access the Embedded Web Server, take these 4 steps:
1. Open a standard Web-browsing application such as Microsoft™ Internet Explorer™ (Version
6.0 and higher) or Netscape™ Navigator™ (Version 7.0 and higher).
2. In the Uniform Resource Locator (URL) field, specify the IP address of the MP-1xx (e.g.,
http://10.1.10.10); the Embedded Web Server’s ‘Enter Network Password’ screen appears, shown in Figure
5-1.
Figure
5-1: Embedded Web Server Login Screen
3. In the ‘User Name’ and ‘Password’ fields, enter the username (default: “Admin”) and
password (default: “Admin”). Note that the username and password are case-sensitive.
4. Click the OK button; the ‘Quick Setup’ screen is accessed (shown in Figure
4-1).
5.6.1 Using Internet Explorer to Access the Embedded Web Server
Internet explorer’s security settings may block access to the gateway’s Web browser if they’re configured incorrectly. In this case, the following message is displayed:
Unauthorized
Correct authorization is required for this area. Either your browser does not perform authorization or your authorization has failed. RomPager server.
To troubleshoot blocked access to Internet Explorer™, take these 2
steps
1. Delete all cookies from the Temporary Internet files. If this does not clear up the problem, the
security settings may need to be altered (refer to Step 2).
2. In Internet Explorer, Tools, Internet Options select the Security tab, and then select Custom
Level. Scroll down until the Logon options are displayed and change the setting to Prompt for username and password and then restart the browser. This fixes any issues related to domain use logon policy.
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5.7 Getting Acquainted with the Web Interface
Figure 5-2 shows the general layout of the Web Interface screen.
Figure
5-2: MP-1xx Web Interface
The Web Interface screen features the following components:
Title bar - contains three configurable elements: corporate logo, a background image and
the product’s name. For information on how to modify these elements refer to
Appendix E on
page 213.
Main menu bar - always appears on the left of every screen to quickly access parameters,
submenus, submenu options, functions and operations.
Submenu bar - appears on the top of screens and contains submenu options.
Main action frame - the main area of the screen in which information is viewed and
configured.
Corporate logo – AudioCodes’ corporate logo. For information on how to remove this logo
Appendix E on page 213.
Control Protocol – the MP-1xx control protocol.
5.7.1 Main Menu Bar
The main menu bar of the Web Interface is divided into the following 7 menus:
Quick Setup – Use this menu to configure the gateway’s basic settings; for the full list of
configurable parameters go directly to ‘Protocol Management’ and ‘Advanced Configuration’ menus. An example of the Quick Setup configuration is described in Section
4.2 on page 36.
Protocol Management – Use this subdivided menu to configure the gateway’s control
protocol parameters and tables (refer to Section
5.8 on page 44).
Main Menu
Ba
r
Corporate
Lo
g
o
Submenu
Ba
r
Title Bar
Main Action
Frame
Control
Protocol
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Advanced Configuration – Use this subdivided menu to set the gateway’s advanced
configuration parameters (for advanced users only) (refer to Section 5.9 on page 103).
Status & Diagnostics – Use this subdivided menu to view and monitor the gateway’s
channels, Syslog messages, hardware / software product information, and to assess the gateway’s statistics and IP connectivity information (refer to Section
5.10 on page 124).
Software Update – Use this subdivided menu when you want to load new software or
configuration files onto the gateway (refer to Section
5.11 on page 131).
Save Configuration – Use this menu to save configuration changes to the non-volatile flash
memory (refer to Section
5.12 on page 139).
Reset – Use this menu to remotely reset the gateway. Note that you can choose to save the
gateway configuration to flash memory before reset (refer to Section
5.12 on page 139).
When positioning your curser over a parameter name (or a table) for more than 1 second, a short description of this parameter is displayed. Note that those parameters that are preceded with an exclamation mark (!) are Not changeable on-the-fly and require reset.
5.7.2 Saving Changes
To save changes to the volatile memory (RAM) press the Submit button (changes to parameters with on-the-fly capabilities are immediately available, other parameter are updated only after a gateway reset). Parameters that are only saved to the volatile memory revert to their previous settings after hardware reset. When performing a software reset (i.e., via Web or SNMP) you can choose to save the changes to the non-volatile memory. To save changes so they are available after a power fail, you must save the changes to the non-volatile memory (flash). When Save Configuration is performed, all parameters are saved to the flash memory.
To save the changes to flash, take these 2 steps:
1. Click the Save Configuration button; the ‘Save Configuration to Flash Memory’ screen
appears.
2. Click the Save Configuration button in the middle of the screen; a confirmation message
appears when the save is complete.
Note: When you reset the MP-1xx from the Web Interface, you can choose to save the configuration to flash memory.
5.7.3 Entering Phone Numbers in Various Tables
Phone numbers entered into various tables on the gateway, such as the Tel to IP routing table, must be entered without any formatting characters. For example, if you wish to enter the phone number 555-1212, it must be entered as 5551212 without the hyphen (-). If the hyphen is entered, the entry does not work. The hyphen character is used in number entry only, as part of a range definition. For example, the entry [20-29] means “all numbers in the range 20 to 29”.
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5.8 Protocol Management
Use this subdivided menu to configure the gateway’s SIP parameters and tables.
Note: Those parameters contained within square brackets are the names used to
configure the parameters via the ini file.
5.8.1 Protocol Definition Parameters
Use this submenu to configure the gateway’s specific SIP protocol parameters.
5.8.1.1 General Parameters
Use this screen to configure general SIP parameters.
To configure the general parameters under Protocol Definition, take
these 4 steps:
1. Open the ‘General Parameters’ screen (Protocol Management menu > Protocol Definition
submenu > General Parameters option); the ‘General Parameters’ screen is displayed.
Figure
5-3: Protocol Definition, General Parameters Screen
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2. Configure the general parameters under Protocol Definition according to Table 5-1.
3. Click the Submit button to save your changes.
4. To save the changes so they are available after a power fail refer to Section
5.12 on page
139.
Table
5-1: Protocol Definition, General Parameters (continues on pages 45 to 47)
Parameter Description
PRACK Mode [PRACKMode]
PRACK mechanism mode for 1XX reliable responses: Disable [0]. Supported [1] (default). Required [2].
Note 1: The Supported and Required headers contain the “100rel” parameter. Note 2: MP-1xx sends PRACK message if 180/183 response is received with “100rel” in
the Supported or the Required headers.
Channel Select Mode [ChannelSelectMode]
Port allocation algorithm for IP to Tel calls. You can select one of the following methods:
By phone number [0] = Select the gateway port according to the called number
(called number is defined in the ‘Endpoint Phone Number’ table).
Cyclic Ascending [1] = Select the next available channel in an ascending cycle
order. Always select the next higher channel number in the hunt group. When the gateway reaches the highest channel number in the hunt group, it selects the lowest channel number in the hunt group and then starts ascending again.
Ascending [2] = Select the lowest available channel. Always start at the lowest
channel number in the hunt group and if that channel is not available, select the next higher channel.
Cyclic Descending [3] = Select the next available channel in descending cycle
order. Always select the next lower channel number in the hunt group. When the gateway reaches the lowest channel number in the hunt group, it selects the highest channel number in the hunt group and then starts descending again.
Descending [4] = Select the highest available channel. Always start at the highest
channel number in the hunt group and if that channel is not available, select the next lower channel.
Number + Cyclic Ascending [5] = First select the gateway port according to the
called number (called number is defined in the ‘Endpoint Phone Number’ table). If the called number isn’t found, then select the next available channel in ascending cyclic order. Note that if the called number is found, but the port associated with this number is busy, the call is released.
The default method is ‘By Phone Number’.
Enable Early Media [EnableEarlyMedia]
No [0] = Early Media is disabled (default). Yes [1] = Enable Early Media. If enabled, the gateway sends 183 Session Progress response with SDP (instead of 180 Ringing), allowing the media stream to be set up prior to the answering of the call.
Note that to send 183 response you must also set the parameter ‘ProgressIndicator2IP’ to 1. If it is equal to 0, 180 Ringing response is sent. Note: Generally, this parameter is set to 1.
Session-Expires Time [SIPSessionExpires]
Determines the timeout (in seconds) for keeping a "re-INVITE" message alive within a SIP session. The SIP session is refreshed (using INVITE) each time this timer expires. The default is 0 (not activated).
Minimum Session-Expires [MINSE]
Defines the time (in seconds) that is used in the Min-SE header field. This field defines the minimum time that the user agent supports for session refresh. The valid range is 10 to 100000. The default value is 90.
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Table 5-1: Protocol Definition, General Parameters (continues on pages 45 to 47)
Parameter Description
Asserted Identity Mode
[AssertedIdMode]
Disable [0] = None (default). Adding PAsserted Identity [1]. Adding PPrefered Identity [2].
The Asserted ID mode defines the header that is used in the generated INVITE request. The header also depends on the calling Privacy: allowed or restricted. The P-asserted (or P-preferred) headers are used to present the originating party’s Caller ID. The Caller ID is composed of a Calling Number and (optionally) a Calling Name. P-asserted (or P-preferred) headers are used together with the Privacy header. If Caller ID is restricted the “Privacy: id” is included. Otherwise for allowed Caller ID the “Privacy: none” is used. If Caller ID is restricted (received from Tel or configured in the gateway), the From header is set to <anonymous@anonymous.invalid>.
Enable T.38 Fax Relay [IsFaxUsed]
Determines the SIP signaling method used to establish and convey a fax session after a fax is detected. No Fax [0] = No fax negotiation using SIP signaling (default). T.38 Relay [1] = Initiates T.38 fax relay. G.711 Transport [2] = Initiates fax using the coder G.711 A-law/µ-law (if not previously selected) with adaptations (refer to note 1). Note 1: Fax adaptations: Echo Canceller = On Silence Compression = Off Echo Canceller Non-Linear Processor Mode = Off Dynamic Jitter Buffer Minimum Delay = 40 Dynamic Jitter Buffer Optimization Factor = 13 Note 2: If the gateway initiates a fax session using G.711 (option 2), a ‘gpmd’ attribute is added to the SDP in the following format: For A-law: ‘a=gpmd:0 vbd=yes;ecan=on’. For µ-law: ‘a=gpmd:8 vbd=yes;ecan=on’. Note 3: When ‘IsFaxUsed’ is set to 1 or 2, the parameter ‘FaxTransportMode’ is ignored.
Detect Fax on Answer Tone [DetFaxOnAnswerTone]
Initiate T.38 on Preamble [0] = Terminating fax gateway initiates T.38 session on receiving of HDLC preamble signal from fax (default) Initiate T.38 on CED [1] = Terminating fax gateway initiates T.38 session on receiving of CED answer tone from fax. Note: This parameters is applicable only if ‘IsFaxUsed = 1’.
SIP Local Port
[LocalSIPPort]
Local UDP port used to receive SIP messages. The default value is 5060.
SIP Destination Port [SIPDestinationPort]
SIP UDP destination port for sending SIP messages. The default value is 5060.
Use “user=phone” in SIP URL [IsUserPhone]
No [0] = "user=phone" string isn’t used in SIP URL. Yes [1] = "user=phone" string is part of the SIP URL (default).
Use “user=phone” in From header [IsUserPhoneInFrom]
No [0] = Doesn’t use ";user=phone" string in From header (default). Yes [1] = ";user=phone" string is part of the From header.
Tel to IP No Answer Timeout [IPAlertTimeout]
Defines the time (in seconds) the gateway waits for a 200 OK response from the called party (IP side) after sending an Invite message. If the timer expires, the call is released. The valid range is 0 to 3600. The default value is 180.
Add Number Plan and Type to Remote Party ID Header [AddTON2RPI]
No [0] = TON/PLAN parameters aren’t included in the RPID header. Yes [1] = TON/PLAN parameters are included in the RPID header (default). If RPID header is enabled (EnableRPIHeader = 1) and ‘AddTON2RPI=1’, it is possible to configure the calling and called number type and number plan using the Number Manipulation tables for TelIP calls.
Use Source Number as Display Name
[UseSourceNumberAsDispl ayName]
No [0] = Interworks the Tel calling name to SIP Display Name (default). Yes [1] = Set Display Name to Calling Number if not configured.
Applicable to TelIP calls. If enabled and calling party name is not defined (CallerDisplayInfoX = <name> is not specified per gateway’s x port), the calling number is used instead.
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Table 5-1: Protocol Definition, General Parameters (continues on pages 45 to 47)
Parameter Description
Play Ringback Tone to IP [PlayRBTone2IP]
Don’t Play [0] = Ringback tone isn’t played to the IP side of the call (default). Play [1] = Ringback tone is played to the IP side of the call after SIP 183 session progress response is sent.
Note 1: To enable the gateway to send a 183 response, set ‘EnableEarlyMedia’ to 1. Note 2: If ‘EnableDigitDelivery = 1’, the gateway doesn’t play a Ringback tone to IP and
doesn’t send a 183 response.
Play Ringback Tone to Tel [PlayRBTone2Tel]
Don’t Play [0] = Ringback Tone isn’t played. Always Play [1] = Ringback Tone is played to the Tel side of the call when 180/183 response is received. Play According to PI [3] = N/A. Play According to 180/183 [2] = Ringback Tone is played to the Tel side of the call if no SDP is received in 180/183 responses. If 180/183 with SDP message is received, the gateway cuts through the voice channel and doesn’t play Ringback tone (default).
Retransmission Parameters
SIP T1 Retransmission Timer [msec] [SipT1Rtx]
The time interval (in msec) between the first transmission of a SIP message and the first retransmission of the same message. The default is 500.
Note: The time interval between subsequent retransmissions of the same SIP message starts with SipT1Rtx and is multiplied by two until SipT2Rtx.
For example (assuming that SipT1Rtx = 500 and SipT2Rtx = 4000): The first retransmission is sent after 500 msec. The second retransmission is sent after 1000 (2*500) msec. The third retransmission is sent after 2000 (2*1000) msec. The fourth retransmission and subsequent retransmissions until SIPMaxRtx are sent after 4000 (2*2000) msec.
SIP T2 Retransmission Timer [msec] [SipT2Rtx]
The maximum interval (in msec) between retransmissions of SIP messages. The default is 4000.
Note: The time interval between subsequent retransmissions of the same SIP message starts with SipT1Rtx and is multiplied by two until SipT2Rtx.
SIP Maximum Rtx [SIPMaxRtx]
Number of UDP retransmissions of SIP messages. The range is 1 to 7. The default value is 7.
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5.8.1.2 Proxy & Registration Parameters
Use this screen to configure parameters that are associated with Proxy and Registration.
To configure the Proxy & Registration parameters, take these 4 steps:
1. Open the ‘Proxy & Registration’ parameters screen (Protocol Management menu >
Protocol Definition submenu > Proxy & Registration option); the ‘Proxy & Registration’
parameters screen is displayed.
Figure
5-4: Proxy & Registration Parameters Screen
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2. Configure the Proxy & Registration parameters according to Table 5-2.
3. Click the Submit button to save your changes, or click the Register or Un-Register buttons
to save your changes and to register / unregister to a Proxy / Registrar.
4. To save the changes so they are available after a power fail refer to Section
5.12 on page
139.
Table
5-2: Proxy & Registration Parameters (continues on pages 49 to 52)
Parameter Description
Enable Proxy [IsProxyUsed]
Don’t Use Proxy [0] = Proxy isn’t used, the internal routing table is used instead (default). Use Proxy [1] = Proxy is used. If you are using a Proxy server, enter the IP address of the primary Proxy server in the Proxy IP address field. If you are not using a Proxy server, you must configure the Tel to IP Routing table on the gateway (described in Section
5.8.4.2 on page 75).
Proxy Name [ProxyName]
Defines the Home Proxy Domain Name. If specified, the Proxy Name is used as Request-URI in REGISTER, INVITE and other SIP messages. If not specified, the Proxy IP address is used instead.
Proxy IP Address [ProxyIP]
IP address (and optionally port number) of the primary Proxy server you are using. Enter the IP address as FQDN or in dotted format notation (for example 201.10.8.1). You can also specify the selected port in the format: <IP Address>:<port>.
This parameter is applicable only if you select ‘Yes’ in the ‘Is Proxy Used’ field. If you enable Proxy Redundancy (by setting EnableProxyKeepAlive=1), the gateway can work with up to three Proxy servers. If there is no response from the primary Proxy, the gateway tries to communicate with the redundant Proxies. When a redundant Proxy is found, the gateway either continues working with it until the next failure occurs or reverts to the primary Proxy (refer to the ‘Redundancy Mode’ parameter). If none of the Proxy servers respond, the gateway goes over the list again.
The gateway also provides real time switching (hotswap mode), between the primary and redundant proxies (‘IsProxyHotSwap=1’). If the first Proxy doesn’t respond to Invite message, the same Invite message is immediately sent to the second Proxy. Note 1: If ‘EnableProxyKeepAlive=1’, the gateway monitors the connection with the Proxies by using keep-alive messages ("OPTIONS"). Note 2: To use Proxy Redundancy, you must specify one or more redundant Proxies using multiple ’ProxyIP= <IP address>’ definitions. Note 3: When port number is specified (e.g., domain.com:5080), DNS SRV queries aren’t performed, even if ‘EnableProxySRVQuery’ is set to 1.
Gateway Name [SIPGatewayName]
Use this parameter to assign a name to the device (For example: ‘gateway1.com’). Ensure that the name you choose is the one that the Proxy is configured with to identify your media gateway. Note: If specified, the gateway Name is used as the host part of the SIP URL, in both ‘To’ and ‘From’ headers. If not specified, the gateway IP address is used instead (default).
Gateway Registration Name
[GWRegistrationName]
Defines the user name that is used in From and To headers of Register messages. Applicable only to single registration per gateway (’AuthenticationMode = 1). If ‘GWRegistrationName’ isn’t specified (default), the ’Username’ parameter is used instead. Note: If ‘“AuthenticationMode=0’, all the gateway’s endpoints are registered with a user name that equals to the endpoint’s phone number.
First Redundant Proxy IP Address [ProxyIP]
IP addresses of the first redundant Proxy you are using. Enter the IP address as FQDN or in dotted format notation (for example 192.10.1.255). You can also specify the selected port in the format: <IP Address>:<port>.
Note 1: This parameter is available only if you select “Yes” in the ‘Enable Proxy’ field. Note 2: When port number is specified, DNS SRV queries aren’t performed, even if
‘EnableProxySRVQuery’ is set to 1. ini file note: The IP address of the first redundant Proxy is defined by the second repetition of the ini file parameter ‘ProxyIP’.
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Table 5-2: Proxy & Registration Parameters (continues on pages 49 to 52)
Parameter Description
Second Redundant Proxy IP Address [ProxyIP]
IP addresses of the second redundant Proxy you are using. Enter the IP address as FQDN or in dotted format notation (for example 192.10.1.255). You can also specify the selected port in the format: <IP Address>:<port>.
Note 1: This parameter is available only if you select “Yes” in the ‘Enable Proxy’ field. Note 2: When port number is specified, DNS SRV queries aren’t performed, even if
‘EnableProxySRVQuery’ is set to 1. ini file note: The IP address of the second redundant Proxy is defined by the third repetition of the ini file parameter ‘ProxyIP’.
Third Redundant Proxy IP Address [ProxyIP]
IP addresses of the third redundant Proxy you are using. Enter the IP address as FQDN or in dotted format notation (for example 192.10.1.255). You can also specify the selected port in the format: <IP Address>:<port>.
Note 1: This parameter is available only if you select “Yes” in the ‘Enable Proxy’ field. Note 2: When port number is specified, DNS SRV queries aren’t performed, even if
‘EnableProxySRVQuery’ is set to 1. ini file note: The IP addresses of the third redundant Proxy is defined by the forth repetition of the ini file parameter ‘ProxyIP’.
Enable Proxy SRV Queries
[EnableProxySRVQuery]
Enables the use of DNS Service Record (SRV) queries to discover Proxy servers. Disable [0] = Disabled (default). Enable [1] = Enabled.
If enabled and the Proxy IP address parameter contains a domain name without port definition (e.g., ProxyIP = domain.com), an SRV query is performed. The SRV query returns up to four Proxy host names and their weights. The gateway then performs DNS A-record queries for each Proxy host name (according to the received weights) to locate up to four Proxy IP addresses. Therefore, if the first SRV query returns two domain names, and the A-record queries return 2 IP addresses each, no more searches are performed. If the Proxy IP address parameter contains a domain name with port definition (e.g., ProxyIP = domain.com:5080), the gateway performs a regular DNS A-record query. Note: This mechanism is applicable only if ‘EnableProxyKeepAlive = 1’.
Redundancy Mode [ProxyRedundancyMode]
Parking [0] = Gateway continues working with the last active Proxy until the next failure (default). Homing [1] = Gateway always tries to work with the primary Proxy server (switches back to the main Proxy whenever it is available). Note: To use Redundancy Mode, enable Keep-alive with Proxy option (Enable Proxy Keep Alive = Yes).
Is Proxy Trusted
[IsTrustedProxy]
This parameter isn’t applicable and must always be set to ‘Yes’ [1]. The parameter ‘AssertedIdMode’ should be used instead.
Enable Registration [IsRegisterNeeded]
No [0] = Gateway doesn’t register to Proxy / Registrar (default). Yes [1] = Gateway registers to Proxy / Registrar when the device is powered up and every Registration Time seconds. Note: The gateway sends a register request for each channel or for the entire gateway (according to the Authentication Mode parameter).
Registrar Name
[RegistrarName]
Registrar Domain Name. If specified, the name is used as Request-URI in Register messages. If isn’t specified (default), the Registrar IP address or Proxy name or Proxy IP address is used instead.
Registrar IP Address [RegistrarIP]
IP address of Registrar server (optional). Enter the IP address in dotted format notation, for example 201.10.8.1. Note: If not specified, the Register request is sent to the primary Proxy server (refer to ‘Proxy IP address’ parameter).
Registration Time [RegistrationTime]
Time (in seconds) for which registration to a Proxy server is valid. The value is used in the "Expires = " header. Typically a value of 3600 is assigned, for one hour registration. The gateway resumes registration when half the defined timeout period expires. The default is 3600 seconds.
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Table 5-2: Proxy & Registration Parameters (continues on pages 49 to 52)
Parameter Description
Re-registration Timing (%) [RegistrationTimeDivider]
Defines the re-registration timing (in percentage). The timing is a percentage of the re­register timing set by the Registration server. The valid range is 50 to 100. The default value is 50. For example: If ‘RegistrationTimeDivider = 70’ (%) and Registration Expires time = 3600, the gateway resends its registration request after 3600 x 70% = 2520 sec.
Registration Retry Time
[RegistrationRetryTime]
Defines the time period (in seconds) after which a Registration request is resent if registration fails with 4xx, or there is no response from the Proxy/Registrar. The default is 30 seconds. The range is 10 to 3600.
Enable Proxy Keep Alive [EnableProxyKeepAlive]
No [0] = Disable (default). Yes [1] = Keep alive with Proxy is enabled. If enabled, ‘OPTIONS’ SIP message is sent every ‘Proxy Keep-Alive Time’. Note: This parameter must be enabled when Proxy redundancy is used.
Proxy Keep Alive Time [ProxyKeepAliveTime]
Defines the Proxy keep-alive time interval (in seconds) between OPTIONS messages. The default value is 60 seconds.
Use Gateway Name for OPTIONS
[UseGatewayNameForOpt ions]
No [0] = Use the gateway’s IP address in keep-alive OPTIONS messages (default). Yes [1] = Use ‘GatewayName’ in keep-alive OPTIONS messages. The OPTIONS Request-URI host part contains either the gateway’s IP address or a string defined by the parameter ‘Gatewayname’. The gateway uses the OPTIONS request as a keep-alive message to its primary and redundant Proxies.
Enable Fallback to Routing Table [IsFallbackUsed]
No [0] = Gateway fallback is not used (default). Yes [1] = Internal Tel to IP Routing table is used when Proxy servers are not available. When the gateway falls back to the internal Tel to IP Routing table, the gateway continues scanning for a Proxy. When the gateway finds an active Proxy, it switches from internal routing back to Proxy routing.
Note: To enable the redundant Proxies mechanism set ‘EnableProxyKeepAlive’ to 1.
PreferRouteTable
[Prefer Routing Table]
Determines if the local Tel to IP routing table takes precedence over a Proxy for routing calls. No [0] = Only Proxy is used to route calls (default). Yes [1] = The Proxy checks the 'Destination IP Address' field in the 'Tel to IP Routing' table for a match with the outgoing call. Only if a match is not found, a Proxy is used. Note: Applicable only if Proxy is not always used (‘AlwaysSendToProxy’ = 0, ‘SendInviteToProxy’ = 0).
Use Routing Table for Host Names and Profiles [AlwaysUseRouteTable]
Use the internal Tel to IP routing table to obtain the URL Host name and (optionally) an IP profile (per call), even if Proxy server is used. No [0] = Don’t use (default). Yes [1] = Use. Note: This Domain name is used, instead of Proxy name or Proxy IP address, in the INVITE SIP URL.
Always Use Proxy [AlwaysSendToProxy]
No [0] = Use standard SIP routing rules (default). Yes [1] = All SIP messages and Responses are sent to Proxy server. Note: Applicable only if Proxy server is used.
Send All Invite to Proxy
[SendInviteToProxy]
No [0] = Invite messages, generated as a result of Transfer or Redirect, are sent directly to the URL (according to the refer-to header in the REFER message or contact header in 30x response) (default). Yes [1] = All Invite messages, including those generated as a result of Transfer or Redirect are sent to Proxy. Note: Applicable only if Proxy server is used and “AlwaysSendtoProxy=0”.
Enable Proxy Hot-Swap [IsProxyHotSwap]
Enable Proxy Hot-Swap redundancy mode. No [0] = Disabled (default). Yes [1] = Enabled. If Hot Swap is enabled, SIP Invite message is first sent to the primary Proxy server. If there is no response from the primary Proxy server for ‘Number of RTX before Hot-Swap’ retransmissions, the Invite message is resent to the redundant Proxy server.
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Table 5-2: Proxy & Registration Parameters (continues on pages 49 to 52)
Parameter Description
Number of RTX Before Hot­Swap [ProxyHotSwapRtx]
Number of retransmitted Invite messages before call is routed (hot swapped) to another Proxy. The range is 1-30. The default is 3. Note: This parameter is also used for alternative routing using the Tel to IP Routing table. If a domain name in the routing table is resolved into 2 IP addresses, and if there is no response for ‘ProxyHotSwapRtx’ retransmissions to the Invite message that is sent to the first IP address, the gateway immediately initiates a call to the second IP address.
User Name
[UserName] Note: The Authentication
table can be used instead.
Username used for Registration and for BASIC/DIGEST authentication process with Proxy. Parameter doesn’t have a default value (empty string). Note: Applicable only if single gateway registration is used (‘Authentication Mode = Authentication Per gateway’).
Password [Password]
Password used for BASIC/DIGEST authentication process with Proxy. Single password is used for all gateway ports. The default is “
Default_Passwd”.
Note: The Authentication table can be used instead.
Cnonce [Cnonce]
String used by the server and client to provide mutual authentication. (Free format i.e., “Cnonce = 0a4f113b”). The default is “Default_Cnonce”.
Authentication Mode [AuthenticationMode]
Per Endpoint [0] = Registration & Authentication separately for each endpoint (default). Per gateway [1] = Single Registration & Authentication for the gateway. Per Ch. Select Mode [2] = N/A. Usually Authentication on a per endpoint basis is used for FXS gateways, in which each endpoint registers (and authenticates) separately with its own username and password. Single Registration and Authentication (Authentication Mode=1) is usually defined for FXO gateways.
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5.8.1.3 Coders
From the Coders screen you can configure the first to fifth preferred coders (and their corresponding ptimes) for the gateway. The first coder is the highest priority coder and is used by the gateway whenever possible. If the far end gateway cannot use the coder assigned as the first coder, the gateway attempts to use the next coder and so forth.
To configure the Gateway’s coders, take these 6 steps:
1. Open the ‘Coders’ screen (Protocol Management menu > Protocol Definition submenu >
Coders option); the ‘Coders’ screen is displayed.
Figure
5-5: Coders Screen
2. From the coder drop-down list, select the coder you want to use. For the full list of available
coders and their corresponding ptimes refer to Table
5-3.
Note: Each coder can appear only once.
3. From the drop-down list to the right of the coder list, select the size of the Voice Packet
(ptime) used with this coder in milliseconds. Selecting the size of the packet determines how many coder payloads are combined into one RTP (voice) packet.
Note 1: The ptime packetization period depends on the selected coder name. Note 2: If not specified, the ptime gets a default value. Note 3: The ptime specifies the maximum packetization time the gateway can receive.
4. Repeat steps 2 and 3 for the second to fifth coders (optional).
5. Click the Submit button to save your changes.
6. To save the changes so they are available after a power fail refer to Section
5.12 on page
139.
Note: Only the ptime of the first coder in the defined coder list is declared in
Invite/200 OK SDP, even if multiple coders are defined.
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Table 5-3: ini File Coder Parameter
Parameter Description CoderName
Enter the coders in the format: CoderName=<Coder>,<ptime>. For example: CoderName = g711Alaw64k,20 CoderName = g711Ulaw64k,40 CoderName = g7231,90
Note 1: This parameter (CoderName) can appear up to 10 times. Note 2: The coder name is case-sensitive.
You can select the following coders: g711Alaw64k – G.711 A-law. g711Ulaw64k – G.711 µ-law. g7231 – G.723.1 6.3 kbps (default). g7231r53 – G.723.1 5.3 kbps. g726 – G.726 ADPCM 32 kbps (Payload Type = 35). g729 G.729A.
Note: G.729B is supported if the coder G.729 is selected and ‘EnableSilenceCompression’ equals 1 or 2.
The RTP packetization period (ptime, in msec) depends on the selected coder name, and can have the following values:
g711 – 10, 20, 30, 40, 50, 60, 80, 100, 120 (default=20). g729 – 10, 20, 30, 40, 50, 60, 80, 100, 120 (default=20). g723 – 30, 60, 90, 120, 150 (default = 30). G.726 – 10, 20, 40, 60, 80, 100, 120 (default=20).
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5.8.1.4 DTMF & Dialing Parameters
Use this screen to configure parameters that are associated with DTMF and dialing.
To configure the dialing parameters, take these 4 steps:
1. Open the ‘DTMF & Dialing’ screen (Protocol Management menu > Protocol Definition
submenu > DTMF & Dialing option); the ‘DTMF & Dialing’ parameters screen is displayed.
Figure
5-6: DTMF & Dialing Parameters Screen
2. Configure the DTMF & Dialing parameters according to Table
5-4.
3. Click the Submit button to save your changes.
4. To save the changes so they are available after a power fail refer to Section
5.12 on page
139.
Table
5-4: DTMF & Dialing Parameters (continues on pages 55 to 57)
Parameter Description
Max Digits in Phone Num
[MaxDigits] Note: Digit Mapping Rules
can be used instead.
Maximum number of digits that can be dialed. You can enter a value from 1 to 49. The default value is 5. Note: Dialing ends when the maximum number of digits is dialed, the Interdigit Timeout expires, the '#' key is dialed, or a digit map pattern is matched.
Inter Digits Timeout [sec] [TimeBetweenDigits]
Time in seconds that the gateway waits between digits dialed by the user. When the Interdigit Timeout expires, the gateway attempts to dial the digits already received. You can enter a value of 1 to 10 seconds. The default value is 4 seconds.
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Table 5-4: DTMF & Dialing Parameters (continues on pages 55 to 57)
Parameter Description
Use Out-of-Band DTMF
[IsDTMFUsed]
Use out-of-band signaling to relay DTMF digits. No [0] = DTMF digits are sent in-band (default). Yes [1] = DTMF digits are sent out-of-band according to the parameter ‘Out-of-band DTMF format’.
Note: When out-of-band DTMF transfer is used (Enable DTMF = Yes), the parameter ‘DTMF Transport Type’ is automatically set to 0 (erase the DTMF digits from the RTP stream).
Out-of-Band DTMF Format [OutOfBandDTMFFormat]
The exact method to send out-of-band DTMF digits. Info (Nortel) [1] = Sends DTMF digits according with "IETF draft-choudhuri-sip-info­digit-00". Info (Cisco) [2] = Sends DTMF digits according with Cisco format (default). Notify (3Com) [3] = NOTIFY format <draft-mahy-sipping-signaled-digits-01.txt>.
Note 1: To use out-of-band DTMF, set ‘Enable DTMF = yes’ (‘IsDTMFUsed=1’). Note 2: When using out-of-band DTMF, the “DTMFTransportType” parameter is
automatically set to 0, to erase the DTMF digits from the RTP stream.
Declare RFC 2833 in SDP
[RxDTMFOption]
Defines the supported Receive DTMF negotiation method. No [0] = Don’t declare RFC 2833 Telephony-event parameter in SDP Yes [3] = Declare RFC 2833 Telephony-event parameter in SDP (default)
The MP-1xx is designed to always be receptive to RFC 2833 DTMF relay packets. Therefore, it is always correct to include the “Telephony-event” parameter as a default in the SDP. However some gateways use the absence of the “telephony-event” from the SDP to decide to send DTMF digits in-band using G.711 coder, if this is the case you can set “RxDTMFOption=0”.
DTMF RFC 2833 Negotiation [TxDTMFOption]
No [0] = No negotiation, DTMF digit is sent according to the parameters ‘DTMF Transport Type’ and ‘RFC2833PayloadType’. Yes [4] = Enable RFC 2833 payload type (PT) negotiation (default).
Note 1: This parameter is applicable only if “IsDTMFUsed=0” (out-of-band DTMF is not used). Note 2: If enabled, the gateway:
Negotiates RFC 2833 payload type using local and remote SDPs.
Sends DTMF packets using RFC 2833 PT according to the PT in the received SDP.
Expects to receive RFC 2833 packets with the same PT as configured by the
“RFC2833PayloadType” parameter.
Note 3: If the remote party doesn’t include the RFC 2833 DTMF relay payload type in the SDP, the gateway uses the same PT for send and for receive. Note 4: If TxDTMFOption is set to 0, the RFC 2833 payload type is set according to the parameter ‘RFC2833PayloadType’ for both transmit and receive.
RFC 2833 Payload Type
[RFC2833PayloadType]
The RFC 2833 DTMF relay dynamic payload type. Range: 96 to 99, 106 to 127; Default = 96 The 100, 102 to 105 range is allocated for proprietary usage.
Note 1: Cisco is using payload type 101 for RFC 2833. Note 2: When RFC 2833 payload type (PT) negotiation is used (TxDTMFOption=4), this
payload type is used for the received DTMF packets. If negotiation isn’t used, this payload type is used for receive and for transmit.
Use Info for Hook-Flash [IsHookFlashUsed]
No [0] = INFO message isn’t sent (default). Yes [1] = Proprietary INFO message with hook-flash is sent when hook-flash is detected (FXS). FXO gateways generate a hook-flash signal when INFO message with hook-flash is received.
Note: When either of the supplementary services (Hold, Transfer or Call Waiting) is enabled, hook-flash is used internally, and thus the hook-flash signal isn’t sent via an INFO message.
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Table 5-4: DTMF & Dialing Parameters (continues on pages 55 to 57)
Parameter Description
Digit Mapping Rules [DigitMapping]
Digit map pattern. If the digit string (dialed number) has matched one of the patterns in the digit map, the gateway stops collecting digits and starts to establish a call with the collected number The digit map pattern contains up to 8 options, each up to 22 characters that are separated by a vertical bar (|). Available notations:
[n-m] represents a range of numbers
‘.’ (single dot) represents repetition
‘x’ represents any single digit
‘T’ represents a dial timer (configured by TimeBetweenDigits parameter)
‘S’ should be used when a specific rule, that is part of a general rule, is to be
applied immediately. For example, if you enter the general rule x.T and the specific rule 11x, you should append ‘S’ to the specific rule 11xS.
For example: 11xS|00T|[1-7]xxx|8xxxxxxx|#xxxxxxx|*xx|91xxxxxxxxxx|9011x.T
Dial Tone Duration [sec] [TimeForDialTone]
Time in seconds that the dial tone is played. The default time is 16 seconds. FXS Gateway ports play the dial tone after phone is picked up; while FXO Gateway ports play the dial tone after port is seized in response to ringing.
Note 1: During play of dial tone, the Gateway waits for DTMF digits. Note 2: ‘TimeForDialTone’ is not applicable when Automatic Dialing is enabled.
Hot Line Dial Tone Duration
[HotLineDialToneDuration]
Duration (in seconds) of the Hotline dial tone. If no digits are received during the Hotline dial tone duration, the gateway initiates a call to a preconfigured number (set in the automatic dialing table). The valid range is 0 to 60. The default time is 5 seconds. Applicable to FXS and FXO gateways.
Enable Special Digits
[IsSpecialDigits]
Disable [0] = "*" or "#" terminate number collection (default). Enable [1] = if you want to allow "*" and "#" to be used for telephone numbers dialed by a user or entered for the endpoint telephone number. Note: The # and * can always be used as first digit of a dialed number, even if you select ‘Disable’ for this parameter.
Default Destination Number [DefaultNumber]
Defines the telephone number that the gateway uses if the parameters ‘TrunkGroup_x’ or ’ChannelList‘ don’t include a phone number. The parameter is used as a starting number for the list of channels comprising all hunt groups in the gateway.
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5.8.2 Configuring the Advanced Parameters
Use this submenu to configure the gateway’s advanced control protocol parameters.
5.8.2.1 General Parameters
Use this screen to configure general control protocol parameters.
To configure the general parameters under Advanced Parameters, take
these 4 steps:
1. Open the ‘General Parameters’ screen (Protocol Management menu > Advanced
Parameters submenu > General Parameters option); the ‘General Parameters’ screen is
displayed.
Figure
5-7: Advanced Parameters, General Parameters Screen
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2. Configure the general parameters under ‘Advanced Parameters’ according to Table 5-5.
3. Click the Submit button to save your changes.
4. To save the changes so they are available after a power fail refer to Section
5.12 on page
139.
Table
5-5: Advanced Parameters, General Parameters (continues on pages 59 to 62)
Parameter Description
Signaling DiffServ
[ControlIPDiffServ]
Defines the value of the 'DiffServ' field in the IP header for SIP messages. The valid range is 0 to 63. The default value is 0.
IP Security [SecureCallsFromIP]
No [0] = Gateway accepts all SIP calls (default). Yes [1] = Gateway accepts SIP calls only from IP addresses defined in the Tel to IP routing table. The gateway rejects all calls from unknown IP addresses. For detailed information on the Tel to IP Routing table refer to Section
5.8.4.2 on page
75.
Note: Specifying the IP address of a Proxy server in the Tel to IP Routing table enables the gateway to only accept calls originating in the Proxy server and rejects all other calls.
Filter Calls to IP [FilterCalls2IP]
Don’t Filter [0] = Disabled (default) Filter [1] = Enabled
If the filter calls to IP feature is enabled, then when a Proxy is used, the gateway first checks the TelIP routing table before making a call through the Proxy. If the number is not allowed (number isn’t listed or a Call Restriction routing rule, IP=0.0.0.0, is applied), the call is released.
Enable Digit Delivery to IP
[EnableDigitDelivery2IP]
Disable [0] = Disabled (default). Enable [1] = Enable digit delivery to IP. The digit delivery feature enables sending of DTMF digits to the destination IP address after the TelIP call was answered. To enable this feature, modify the called number to include at least one ’p’ character. The gateway uses the digits before the ‘p’ character in the initial Invite message. After the call was answered the gateway waits for the required time (# of ‘p’ * 1.5 seconds) and then sends the rest of the DTMF digits using the method chosen (in-band, out-of­band).
Note: The called number can include several ‘p’ characters (1.5 seconds pause). For example, the called number can be as follows: pp699, p9p300.
Enable Digit Delivery to Tel
[EnableDigitDelivery]
Disable [0] = Disabled (default). Enable [1] = Enable Digit Delivery feature for MP-1xx/FXO & FXS.
The digit delivery feature enables sending of DTMF digits to the gateway’s port after the line is offhooked (FXS) or seized (FXO). For IPTel calls, after the line is offhooked / seized, the MP-1xx plays the DTMF digits (of the called number) towards the phone line.
Note 1: The called number can also include the characters ‘p’ (1.5 seconds pause) and ‘d’ (detection of dial tone). If the character ‘d’ is used, it must be the first “digit” in the called number. The character ‘p’ can be used several times. For example, the called number can be as follows: d1005, dpp699, p9p300. To add the ‘d’ and ‘p’ digits, use the usual number manipulation rules. Note 2: To use this feature with FXO gateways, configure the gateway to work in one stage dialing mode. Note 3: If the parameter ‘EnableDigitDelivery’ is enabled, it is possible to configure the gateway to wait for dial tone per destination phone number (before or during dialing of destination phone number), therefore the parameter ‘IsWaitForDialTone’ (that is configurable for the entire gateway) is ignored. Note 4: The FXS gateway sends 200 OK messages only after it finishes playing the DTMF digits to the phone line.
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Table 5-5: Advanced Parameters, General Parameters (continues on pages 59 to 62)
Parameter Description
Enable DID Wink [EnableDIDWink]
Disable [0] = DID is disabled (default). Enable [1] = Enable DID. If enabled, the MP-1xx can be used for connection to EIA/TIA-464B DID Loop Start lines. Both FXO (detection) and FXS (generation) are supported. An FXO gateway dials DTMF digits after a Wink signal is detected (instead of a Dial tone). An FXS gateway generates the Wink signal after the detection of offhook (instead of playing a Dial tone).
Reanswer Time [RegretTime]
The time period (in seconds) after user hangs up the phone and before call is disconnected (FXS). Also called regret time. The default time is 0 seconds.
Disconnect and Answer Supervision
Enable Polarity Reversal [EnableReversalPolarity]
Disable [0] = Disable the polarity reversal service (default). Enable [1] = Enable the polarity reversal service. If the polarity reversal service is enabled, then the FXS gateway changes the line polarity on call answer and changes it back on call release. The FXO gateway sends a 200 OK response when polarity reversal signal is detected, and releases a call when a second polarity reversal signal is detected.
Enable Current Disconnect [EnableCurrentDisconnect]
Disable [0] = Disable the current disconnect service (default). Enable [1] = Enable the current disconnect service. If the current disconnect service is enabled, the FXO gateway releases a call when current disconnect signal is detected on its port, while the FXS gateway generates a "Current Disconnect Pulse" after a call is released from IP. The current disconnect duration is determined by the parameter ‘CurrentDisconnectDuration’. The current disconnect threshold (FXO only) is determined by the parameter ‘CurrentDisconnectDefaultThreshold’. The frequency at which the analog line voltage is sampled is determined by the parametr ‘TimeToSampleAnalogLineVoltage’.
Disconnect on Broken Connection
[DisconnectOnBrokenConn ection]
No [0] = Don’t release the call. Yes [1] = Call is released if RTP packets are not received for a predefined timeout (default).
Note 1: If enabled, the timeout is set by the parameter ‘BrokenConnectionEventTimeout’, in 100 msec resolution. The default timeout is 10 seconds: (BrokenConnectionEventTimeout =100). Note 2: This feature is applicable only if RTP session is used without Silence Compression. If Silence Compression is enabled, the Gateway doesn’t detect that the RTP connection is broken. Note 3: During a call, if the source IP address (from where the RTP packets were sent) is changed without notifying the Gateway, the Gateway filters these RTP packets. To overcome this issue, set ‘DisconnectOnBrokenConnection=0’; the Gateway doesn’t detect RTP packets arriving from the original source IP address, and switches (after 300 msec) to the RTP packets arriving from the new source IP address.
Broken Connection Timeout
[BrokenConnectionEventTi meout]
The amount of time (in 100 msec units) an RTP packet isn’t received, after which a call is disconnected. The valid range is 1 to 1000. The default value is 100 (10 seconds).
Note 1: Applicable only if ‘DisconnectOnBrokenConnection = 1’. Note 2: Currently this feature works only if Silence Suppression is disabled.
Disconnect Call on Silence Detection [EnableSilenceDisconnect]
Yes [1] = The FXO gateway disconnect calls in which silence occurs in both (call) directions for more than 120 seconds. No [0] = Call is not disconnected when silence is detected (default).
The silence duration can be set by the ‘FarEndDisconnectSilencePeriod’ parameter (default 120).
Note: To activate this feature set DSP Template to 2 or 3.
Silence Detection Period [sec]
[FarEndDisconnectSilenceP eriod]
Duration of silence period (in seconds) prior to call disconnection. The range is 10 to 28800 (8 hours). The default is 120 seconds. Applicable to gateways, that use DSP templates 2 or 3.
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Table 5-5: Advanced Parameters, General Parameters (continues on pages 59 to 62)
Parameter Description
Silence Detection Method
[FarEndDisconnectSilenceM ethod]
Silence detection method. None [0] = Silence detection option is disabled. Packets Count [1] = According to packet count. Voice/Energy Detectors [2] = According to energy and voice detectors (default). All [3] = According to packet count and energy / voice detectors.
CDR and Debug
CDR Server IP Address
[CDRSyslogServerIP]
Defines the destination IP address for CDR logs.
The default value is a null string that causes the CDR messages to be sent with all Syslog messages. Note: The CDR messages are sent to UDP port 514 (default Syslog port).
CDR Report Level [CDRReportLevel]
None [0] = Call Detail Recording (CDR) information isn’t sent to the Syslog server (default). End Call [1] = CDR information is sent to the Syslog server at end of each Call. Start & End Call [2] = CDR information is sent to the Syslog server at the start and at the end of each Call. The CDR Syslog message complies with RFC 3161 and is identified by: Facility = 17 (local1) and Severity = 6 (Informational).
Debug Level [GwDebugLevel]
Syslog logging level. One of the following debug levels can be selected: 0 [0] = Debug is disabled (default) 1 [1] = Flow debugging is enabled 2 [2] = Flow and device interface debugging are enabled 3 [3] = Flow, device interface and stack interface debugging are enabled 4 [4] = Flow, device interface, stack interface and session manager debugging are enabled 5 [5] = Flow, device interface, stack interface, session manager and device interface expanded debugging are enabled.
Note: Usually set to 5 if debug traces are needed. Misc. Parameters Progress Indicator to IP
[ProgressIndicator2IP]
No PI [0] = For IPTel calls, the gateway sends “180 Ringing” SIP response to IP after
placing a call to phone (FXS) or to PBX (FXO).
PI = 1, PI = 8 [1], [8] = For IPTel calls, if ‘EnableEarlyMedia=1’, the gateway sends
“183 session in progress” message + SDP, immediately after a call is placed to
Phone/PBX. This is used to cut through the voice path, before remote party answers the
call, enabling the originating party to listen to network Call Progress Tones (such as
Ringback tone or other network announcements).
Not Configured [-1] = Default values are used.
The default for FXO gateways is 1; The default for FXS gateways is 0. Enable Busy Out
[EnableBusyOut]
No [0] = ‘Busy out’ feature is not used (default).
Yes [1] = The MP-1xx/FXS gateway plays a reorder tone when the phone is offhooked
and one of the following occurs:
There is a network problem.
Proxy servers do not respond and the internal routing table is not configured. Default Release Cause
[DefaultReleaseCause]
Default Release Cause (to IP) for IPTel calls, used when the gateway initiates a call
release, and if an explicit matching cause for this release isn’t found, a default release
cause can be configured:
The default release cause is: NO_ROUTE_TO_DESTINATION (3).
Other common values are: NO_CIRCUIT_AVAILABLE (34),
DESTINATION_OUT_OF_ORDER (27), etc.
Note: The default release cause is described in the Q.931 notation, and is translated to
corresponding SIP 40x or 50x value. For example: 404 for 3, 503 for 34 and 502 for 27. Delay After Reset [sec]
[GWAppDelayTime]
Defines the amount of time (in seconds) the gateway’s operation is delayed after a reset
cycle.
The valid range is 0 to 600. The default value is 5 seconds.
Note: This feature helps to overcome connection problems caused by some LAN
routers or IP configuration parameters change by a DHCP Server.
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Table 5-5: Advanced Parameters, General Parameters (continues on pages 59 to 62)
Parameter Description
Max Number of Active Calls
[MaxActiveCalls]
Defines the maximum number of calls that the gateway can have active at the same
time. If the maximum number of calls is reached, new calls are not established.
The default value is max available channels (no restriction on the maximum number of
calls). The valid range is 1 to max number of channels. Max Call Duration (sec)
[MaxCallDuration]
Defines the maximum call duration in seconds. If this time expires, both sides of the call
are released (IP and Tel).
The default time is 0 seconds (no limitation). Enable LAN Watchdog
[EnableLanWatchDog]
Disable [0] = Disable LAN Watch-Dog (default).
Enable [1] = Enable LAN Watch-Dog.
If LAN Watch-Dog is enabled, the gateway restarts when a network failure is detected. Enable Calls Cut Through
[CutThrough]
Enables users to receive incoming IP calls while the port is in an offhooked state.
Disable [0] = Disabled (default).
Enable [1] = Enabled.
If enabled, FXS gateways answer the call and “cut through” the voice channel, if there is
no other active call on that port, even if the port is in offhooked state.
When the call is terminated (by the remote party), the gateway plays a reorder tone for
‘TimeForReorderTone’ seconds and is then ready to answer the next incoming call,
without onhooking the phone.
The waiting call is automatically answered by the gateway when the current call is
terminated (EnableCallWaiting=1).
Note: This option is applicable only to FXS gateways.
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5.8.2.2 Supplementary Services
Use this screen to configure parameters that are associated with supplementary services. For detailed information on the supplementary services, refer to Section
8.4 on page 153.
To configure the supplementary services’ parameters, take these 4
steps:
1. Open the ‘Supplementary Services’ screen (Protocol Management menu > Advanced
Parameters submenu > Supplementary Services option); the ‘Supplementary Services’
screen is displayed.
Figure
5-8: Supplementary Services Parameters Screen
2. Configure the supplementary services parameters according to Table
5-6.
3. Click the Submit button to save your changes, or click the Subscribe for MWI or Un-
Subscribe for MWI buttons to save your changes and to subscribe / unsubscribe to the MWI
server.
4. To save the changes so they are available after a power fail refer to Section
5.12 on page
139.
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Table 5-6: Supplementary Services Parameters (continues on pages 64 to 65)
Parameter Description
Enable Hold [EnableHold]
No [0] = Disable the Hold service (default). Yes [1] = Enable the Hold service. If the Hold service is enabled, a user can activate Hold (or Unhold) using the hook-flash. On receiving a Hold request, the remote party is put on-hold and hears the hold tone. Note: To use this service, the gateways at both ends must support this option.
Hold Format
[HoldFormat]
Determines the format of the hold request.
0.0.0.0 [0] = The connection IP address in SDP is 0.0.0.0 (default). Send Only [1] = The last attribute of the SDP contains the following “a=sendonly”.
Enable Transfer [EnableTransfer]
No [0] = Disable the Call Transfer service (default). Yes [1] = Enable the Call Transfer service (using REFER). If the Transfer service is enabled, the user can activate Transfer using hook-flash signaling. If this service is enabled, the remote party performs the call transfer.
Note 1: To use this service, the gateways at both ends must support this option. Note 2: To use this service, set the parameter ‘Enable Hold’ to ‘Yes’.
Transfer Prefix [xferPrefix]
Defined string that is added, as a prefix, to the transferred / forwarded called number, when Refer / Redirect message is received. Note 1: The number manipulation rules apply to the user part of the “REFER-TO / Contact” URL before it is sent in the INVITE message. Note 2: The ‘xferprefix’ parameter can be used to apply different manipulation rules to differentiate the transferred / forwarded number from the original dialed number.
Enable Call Forward [EnableForward]
No [0] = Disable the Call Forward service (default). Yes [1] = Enable Call Forward service (using REFER). For FXS gateways a Call Forward table must be defined to use the Call Forward service. To define the Call Forward table, refer to Section
5.8.8.4 on page 96.
Note: To use this service, the gateways at both ends must support this option.
Enable Call Waiting [EnableCallWaiting]
No [0] = Disable the Call Waiting service (default). Yes [1] = Enable the Call Waiting service.
If enabled, when an FXS gateway receives a call on a busy endpoint, it responds with a 182 response (and not with a 486 busy). The gateway plays a call waiting indication signal. When hook-flash is detected, the gateway switches to the waiting call. The gateway that initiated the waiting call plays a Call Waiting Ringback tone to the calling party after a 182 response is received. Note 1: The gateway’s Call Progress Tones file must include a "call waiting Ringback” tone (caller side) and a "call waiting” tone (called side, FXS only). Note 2: The ‘Enable Hold’ parameter must be enabled on both the calling and the called sides. For information on the Call Waiting feature refer to Section
8.4.5 on page 155.
For information on the Call Progress Tones file refer to Section
7.1 on page 143.
Number of Call Waiting Indications
[NumberOfWaitingIndication s]
Number of waiting indications that are played to the receiving side of the call (FXS only) for Call Waiting. The default value is 2.
Time Between Call Waiting Indications
[TimeBetweenWaitingIndica tions]
Difference (in seconds) between call waiting indications (FXS only) for Call Waiting. The default value is 10 seconds.
Time before Waiting Indication
[TimeBeforeWaitingIndicatio n]
Defines the interval (in seconds) before a call waiting indication is played to the port that is currently in a call (FXS only). The valid range is 0 to 100. The default time is 0 seconds.
[Waiting Beep Duration]
WaitingBeepDuration
Duration (in msec) of waiting indications that are played to the receiving side of the call (FXS only) for Call Waiting. The default value is 300.
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Table 5-6: Supplementary Services Parameters (continues on pages 64 to 65)
Parameter Description
Enable Caller ID [EnableCallerID]
No [0] = Disable the Caller ID service (default). Yes [1] = Enable the Caller ID service. If the Caller ID service is enabled, then, for FXS gateways, calling number and Display text are sent to gateway port. For FXO gateways, the Caller ID signal is detected and is sent to IP in SIP INVITE message (as "Display" element). For information on the Caller ID table refer to Section
5.8.8.3 on page 95.
To disable/enable caller I generation per port, refer to Section 5.8.8.5 on page 98.
Caller ID Type [CallerIDType]
Defines one of the following standards for detection (FXO) and generation (FXS) of Caller ID signals. Bellcore [0] (default). ETSI [1]. NTT [2]. British [4] DTMF ETSI [16] Denmark [17] India [18] Brazil [19] Note: The Caller ID signals are generated/detected between the first and the second rings.
MWI Parameters
Enable MWI
[EnableMWI]
Enable MWI (message waiting indication). Disable [0] = Disabled (default). Enable [1] = MWI service is enabled. This parameter is applicable only to FXS gateways. Note: The MP-1xx only supports reception of MWI. For detailed information on MWI, refer to Section
8.4.6 on page 156.
MWI Analog Lamp
[MWIAnalogLamp]
Disable [0] = Disable (default). Enable [1] = Enable visual Message Waiting Indication, supplies line voltage of approximately 100 VDC to activate the phone’s lamp. This parameter is applicable only to FXS gateways.
MWI Display
[MWIDisplay]
Disable [0] = MWI information isn’t sent to display (default). Enable [1] = MWI information is sent to display.
If enabled, the gateway generates an MWI FSK message that is displayed on the MWI display. This parameter is applicable only to FXS gateways.
Subscribe to MWI
[EnableMWISubscription]
Disable [0] = Disable MWI subscription (default). Enable [1] = Enable subscription to MWI (to MWIServerIP address). Note: Use the parameter ‘SubscriptionMode’ (described in Table
5-26 on page 101) to
determine whether the gateway subscribes separately per endpoint of for the entire gateway.
MWI Server IP Address
[MWIServerIP]
MWI server IP address. If provided, the gateway subscribes to this IP address. Can be configured as a numerical IP address or as a domain name. If not configured, the Proxy IP address is used instead.
MWI Subscribe Expiration Time
[MWIExpirationTime]
MWI subscription expiration time in seconds. The default is 7200 seconds. The range is 10 to 72000.
MWI Subscribe Retry Time
[SubscribeRetryTime]
Subscription retry time in seconds. The default is 120 seconds. The range is 10 to 7200.
Stutter Tone Duration
[StutterToneDuration]
Duration (in msec) of the played stutter dial tone that indicates waiting message(s). The default is 2000 (2 seconds). The range is 1000 to 60000. The Stutter tone is played (instead of a regular Dial tone) when a MWI is received. The tone is composed of a ‘Confirmation tone’ that is played for ‘StutterToneDuration’ followed by a ‘Stutter tone’. Both tones are defined in the CPT file. Note: This parameter is applicable only to FXS gateways. For detailed information on Message Waiting Indication (MWI), refer to Section
8.4.6 on
page 156.
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5.8.2.3 Keypad Features
The Keypad Features screen (applicable only to FXS gateways) enables you to activate / deactivate the following features directly from the connected telephone’s keypad:
Call Forward (refer to Section
5.8.8.4 on page 96).
Caller ID Restriction (refer to Section
5.8.8.3 on page 95).
Hotline (refer to Section
5.8.8.2 on page 94).
To configure the keypad features, take these 4 steps:
1. Open the ‘Keypad Features’ screen (Protocol Management menu > Advanced
Parameters submenu > Keypad Features option); the ‘Keypad Features’ screen is
displayed.
Figure
5-9: Keypad Features Screen
2. Configure the Keypad Features according to Table
5-7.
3. Click the Submit button to save your changes.
4. To save the changes so they are available after a power fail refer to Section
5.12 on page
139.
Note: The method used by the gateway to collect dialed numbers is identical to the
method used during a regular call (i.e., max digits, interdigit timeout, digit map, etc.).
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Table 5-7: Keypad Features Parameters
Parameter Description Forward
Unconditional [KeyCFUnCond]
Keypad sequence that activates the immediate forward option.
No Answer [KeyCFNoAnswer]
Keypad sequence that activates the forward on no answer option.
On Busy [KeyCFBusy]
Keypad sequence that activates the forward on busy option.
On Busy or No Answer [KeyCFBusyOrNoAnswer]
Keypad sequence that activates the forward on ‘busy or no answer’ option.
Do Not Disturb [KeyCFDoNotDisturb]
Keypad sequence that activates the Do Not Disturb option.
To activate the required forward method from the telephone:
Dial the preconfigured sequence number on the keypad; a dial tone is heard.
Dial the telephone number to which the call is forwarded (terminate the number with #); a confirmation tone is
heard.
Deactivate [KeyCFDeact]
Keypad sequence that deactivates any of the forward options. After the sequence is pressed a confirmation tone is heard.
Caller ID Restriction
Activate [KeyCLIR]
Keypad sequence that activates the restricted Caller ID option. After the sequence is pressed a confirmation tone is heard.
Deactivate [KeyCLIRDeact]
Keypad sequence that deactivates the restricted Caller ID option. After the sequence is pressed a confirmation tone is heard.
Hotline
Activate [KeyHotLine]
Keypad sequence that activates the delayed hotline option. To activate the delayed hotline option from the telephone:
Dial the preconfigured sequence number on the keypad; a dial tone is heard.
Dial the telephone number to which the phone automatically dials after a
configurable delay (terminate the number with #); a confirmation tone is heard.
Deactivate [KeyHotLineDeact]
Keypad sequence that deactivates the delayed hotline option. After the sequence is pressed a confirmation tone is heard.
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5.8.3 Configuring the Number Manipulation Tables
The VoIP gateway provides four Number Manipulation tables for incoming and outgoing calls. These tables are used to modify the destination and source telephone numbers so that the calls can be routed correctly.
The Manipulation Tables are:
Destination Phone Number Manipulation Table for IPTel calls
Destination Phone Number Manipulation Table for TelIP call
Source Phone Number Manipulation Table for IPTel calls
Source Phone Number Manipulation Table for TelIP calls
Note: Number manipulation can occur either before or after a routing decision is
made. For example, you can route a call to a specific hunt group according to its original number, and then you can remove / add a prefix to that number before it is routed. To control when number manipulation is done, set the IP
to Tel Routing Mode (described in Table 5-12) and the Tel to IP Routing Mode (described in Table 5-11) parameters.
Possible uses for number manipulation can be as follows:
To strip/add dialing plan digits from/to the number. For example, a user could dial 9 in front
of each number in order to indicate an external line. This number (9) can be removed here before the call is setup.
Allow / disallow Caller ID information to be sent according to destination / source prefixes.
For detailed information on Caller ID refer to Section
5.8.8.3 on page 95.
To configure the Number Manipulation tables, take these 5 steps:
1. Open the Number Manipulation screen you want to configure (Protocol Management menu
> Manipulation Tables submenu); the relevant Manipulation table screen is displayed.
Figure
5-10 shows the ‘Source Phone Number Manipulation Table for TelIP calls’.
Figure
5-10: Source Phone Number Manipulation Table for TelIP calls
2. In the ‘Table Index’ drop-down list, select the range of entries that you want to edit (up to 20
entries can be configured for Source Number Manipulation and 50 entries for Destination Number Manipulation).
3. Configure the Number Manipulation table according to Table
5-8.
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4. Click the Submit button to save your changes.
5. To save the changes so they are available after a power fail refer to Section
5.12 on page
139.
Table
5-8: Number Manipulation Parameters
Parameter Description
Destination Prefix Each entry in the Destination Prefix fields represents a destination telephone number
prefix. An asterisk (*) represents any number.
Source Prefix Each entry in the Source Prefix fields represents a source telephone number prefix.
An asterisk (*) represents any number.
Source IP Each entry in the Source IP fields represents the source IP address of the call
(obtained from the Contact header in the Invite message). This column only applies to the ‘Destination Phone Number Manipulation Table for IP to Tel’. Note: The source IP address can include the “x” wildcard to represent single
digits.
For example: 10.8.8.xx represents all the addresses between 10.8.8.10 to 10.8.8.99.
The manipulation rules are applied to any incoming call whose:
Destination number prefix matches the prefix defined in the ‘Destination Number’ field.
Source number prefix matches the prefix defined in the ‘Source Prefix’ field.
Source IP address matches the IP address defined in the ‘Source IP’ field (if applicable).
Note that number manipulation can be performed using a combination of each of the above criteria, or using each criterion independently. Note: For available notations that represent multiple numbers refer to Section 5.8.3.1 on page 71.
Num of stripped digits
Enter the number of digits that you want to remove from the left of the telephone
number prefix. For example, if you enter 3 and the phone number is 5551234, the new phone number is 1234.
Enter the number of digits (in brackets) that you want to remove from the right of
the telephone number prefix.
Note: A combination of the two options is allowed (e.g., 2(3)).
Prefix / Suffix to add
Prefix - Enter the number / string you want to add to the front of the phone
number. For example, if you enter 9 and the phone number is 1234, the new number is 91234.
Suffix - Enter the number / string (in brackets) you want to add to the end of the
phone number. For example, if you enter (00) and the phone number is 1234, the new number is 123400.
Note: You can enter a prefix and a suffix in the same field (e.g., 9(00)).
Number of digits to leave Enter the number of digits that you want to leave from the right.
Note: The manipulation rules are executed in the following order:
1. Num of stripped digits
2. Number of digits to leave
3. Prefix / suffix to add
Figure
5-10 on the previous page exemplifies the use of these manipulation rules in the ‘Source Phone Number
Manipulation Table for TelIP Calls’:
When destination number equals 035000 and source number equals 20155, the source number is changed to
97220155.
When source number equals 1001876, it is changed to 587623.
Source number 1234510012001 is changed to 20018.
Source number 3122 is changed to 2312.
Presentation Select ‘Allowed’ to send Caller ID information when a call is made using these
destination / source prefixes. Select ‘Restricted’ if you want to restrict Caller ID information for these prefixes. When set to ‘Not Configured’, the privacy is determined according to the Caller ID table (refer to Section 5.8.8.3 on page 95). Note: If ‘Presentation’ is set to ‘Restricted’ and ‘Asserted Identity Mode’ is set to ‘P­Asserted’, the From header in Invite message is: From: “anonymous” <sip: anonymous@anonymous.invalid> and “privacy: id” header is included in the Invite message.
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Table 5-9: Number Manipulation ini File Parameters (continues on pages 70 to 71)
Parameter Description
NumberMapTel2IP
Manipulates the destination number for Tel to IP calls. NumberMapTel2IP = a,b,c,d,e,f,g
a = Destination number prefix b = Number of stripped digits from the left, or (if brackets are used) from the right. A combination of both options is allowed. c = String to add as prefix, or (if brackets are used) as suffix. A combination of both options is allowed. d = Number of remaining digits from the right e = Number Plan used in RPID header f = Number Type used in RPID header g = Source number prefix
The ‘b’ to ‘f’ manipulation rules are applied if the called and calling numbers match the ‘a’ and ‘g’ conditions.
The manipulation rules are executed in the following order: ‘b’, ‘d’ and ‘c’. Parameters can be skipped by using the sign "$$", for example: NumberMapTel2IP=01,2,972,$$,0,0,$$ NumberMaPTel2IP=03,(2),667,$$,0,0,22 Note: Number Plan & Type can optionally be used in Remote Party ID (RPID) header by using the ‘EnableRPIHeader’ and ‘AddTON2RPI’ parameters.
NumberMapIP2Tel
Manipulate the destination number for IP to Tel calls. NumberMapIP2Tel = a,b,c,d,e,f,g,h,i
a = Destination number prefix. b = Number of stripped digits from the left, or (if brackets are used) from the right. A combination of both options is allowed. c = String to add as prefix, or (if brackets are used) as suffix. A combination of both options is allowed. d = Number of remaining digits from the right. e = Not applicable, set to $$. f = Not applicable, set to $$. g = Source number prefix. h = Not applicable, set to $$. i = Source IP address (obtained from the Contact header in the Invite message).
The ‘b’ to ‘d’ manipulation rules are applied if the called and calling numbers match the ‘a’, ‘g’ and ‘i’ conditions.
The manipulation rules are executed in the following order: ‘b’, ‘d’ and ‘c’. Parameters can be skipped by using the sign "$$", for example: NumberMapIP2Tel =01,2,972,$$,$$,$$,034,$$,10.13.77.8 NumberMapIP2Tel =03,(2),667,$$,$$,$$,22 Note: The Source IP address can include the “x” wildcard to represent single
digits.
For example: 10.8.8.xx represents all the addresses between 10.8.8.10 to 10.8.8.99.
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Parameter Description
SourceNumberMapTel2IP
SourceNumberMapTel2IP = a,b,c,d,e,f,g,h
a = Source number prefix b = Number of stripped digits from the left, or (if in brackets are used) from right. A combination of both options is allowed. c = String to add as prefix, or (if in brackets are used) as suffix. A combination of both options is allowed. d = Number of remaining digits from the right e = Number Plan used in RPID header f = Number Type used in RPID header g = Destination number prefix h = Calling number presentation (0 to allow presentation, 1 to restrict presentation)
The ‘b’ to ‘f’ and ‘h’ manipulation rules are applied if the called and calling numbers match the ‘a’ and ‘g’ conditions.
The manipulation rules are executed in the following order: ‘b’, ‘d’ and ‘c’. Parameters can be skipped by using the sign "$$", for example: SourceNumberMapTel2IP=01,2,972,$$,0,0,$$,1 SourceNumberMapTel2IP=03,(2),667,$$,0,0,22 Note 1: ‘Presentation’ is set to ‘Restricted’ only if ‘Asserted Identity Mode’ is set to ‘P-Asserted’. Note 2: Number Plan & Type can optionally be used in Remote Party ID (RPID) header by using the ‘EnableRPIHeader’ and ‘AddTON2RPI’ parameters.
SourceNumberMapIP2Tel
Manipulate the destination number for IP to Tel calls. NumberMapIP2Tel = a,b,c,d,e,f,g
a = Source number prefix b = Number of stripped digits from the left, or (if brackets are used) from the right. A combination of both options is allowed. c = String to add as prefix, or (if brackets are used) as suffix. A combination of both options is allowed. d = Number of remaining digits from the right e = Not in use, should be set to $$ f = Not in use, should be set to $$ g = Destination number prefix
The ‘b’ to ‘d’ manipulation rules are applied if the called and calling numbers match the ‘a’ and ‘g’ conditions.
The manipulation rules are executed in the following order: ‘b’, ‘d’ and ‘c’. Parameters can be skipped by using the sign "$$", for example: NumberMapIP2Tel =01,2,972,$$,$$,$$,034 NumberMapIP2Tel =03,(2),667,$$,$$,$$,22
5.8.3.1 Dialing Plan Notation
The dialing plan notation applies, in addition to the four Manipulation tables, also to TelIP Routing table and to IPHunt Group Routing table.
When entering a number in the destination and source ‘Prefix’ columns, you can create an entry that represents multiple numbers using the following notation:
[n-m] represents a range of numbers
[n,m] represents multiple numbers. Note that this notation only supports single digit numbers.
x represents any single digit
# (that terminates the number) represents the end of a number
A single asterisk (*) represents any number
For example:
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[5551200-5551300]# represents all numbers from 5551200 to 5551300
[2,3,4] represents all numbers that start with the numbers 2, 3 and 4
54324 represents any number that starts with 54324
54324xx# represents a 7 digit number that starts with 54324
123[100-200]# represents all numbers from 123100 to 123200.
The VoIP gateway matches the rules starting at the top of the table. For this reason, enter more specific rules above more generic rules. For example, if you enter 551 in entry 1 and 55 in entry 2, the VoIP gateway applies rule 1 to numbers that starts with 551 and applies rule 2 to numbers that start with 550, 552, 553, 554, 555, 556, 557, 558 and 559. However if you enter 55 in entry 1 and 551 in entry 2, the VoIP gateway applies rule 1 to all numbers that start with 55 including numbers that start with 551.
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5.8.4 Configuring the Routing Tables
Use this submenu to configure the gateway’s IPTel and TelIP routing tables and their associated parameters.
5.8.4.1 General Parameters
Use this screen to configure the gateway’s IPTel and TelIP routing parameters.
To configure the general parameters under Routing Tables, take these 4
steps:
1. Open the ‘General Parameters’ screen (Protocol Management menu > Routing Tables
submenu > General option); the ‘General Parameters’ screen is displayed.
Figure
5-11: Routing Tables, General Parameters Screen
2. Configure the general parameters under ‘Routing Tables’ according to Table
5-10.
3. Click the Submit button to save your changes.
4. To save the changes so they are available after a power fail refer to Section
5.12 on page
139.
Table
5-10: Routing Tables, General Parameters (continues on pages 73 to 74)
Parameter Description
Add Hunt Group ID as Prefix [AddTrunkGroupAsPrefix]
No [0] = Don’t add hunt group ID as prefix (default). Yes [1] = Add hunt group ID as prefix to called number. If enabled, then the hunt group ID is added as a prefix to the destination phone number for TelIP calls.
Note 1: This option can be used to define various routing rules. Note 2: To use this feature you must configure the hunt group IDs.
Add Port Number as Prefix [AddPortAsPrefix]
No [0] = Disable the add port as prefix service (default). Yes [1] = Enable the add port as prefix service. If enabled, then the gateway’s port number (single digit in the range 1 to 8 in MP-10x, two digits in the range 01 to 24 in MP-124) is added as a prefix to the destination phone number for TelIP calls. Note: This option can be used to define various routing rules.
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Table 5-10: Routing Tables, General Parameters (continues on pages 73 to 74)
Parameter Description
IP to Tel Remove Routing Table Prefix [RemovePrefix]
No [0] = Don't remove prefix (default) Yes [1] = Remove the prefix (defined in the IP to Hunt Group Routing table) from a telephone number for an IPTel call, before forwarding it to Tel. For example: To route an incoming IPTel Call with destination number 21100, the IP to Hunt Group Routing table is scanned for a matching prefix. If such prefix is found, 21 for instance, then before the call is routed to the corresponding hunt group the prefix (21) is removed from the original number, so that only 100 is left. Note 1: Applicable only if number manipulation is performed after call routing for IPTel calls (refer to ‘IP to Tel Routing Mode’ parameter). Note 2: Similar operation (of removing the prefix) is also achieved by using the usual number manipulation rules.
Enable Alt Routing Tel to IP [AltRoutingTel2IPEnable]
No [0] = Disable the Alternative Routing feature (default). Yes [1] = Enable the Alternative Routing feature. Status Only [2] = The Alternative Routing feature is disabled. A read only information on the quality of service of the destination IP addresses is provided. For information on the Alternative Routing feature refer to Section
8.3 on page 152.
Alt Routing Tel to IP Mode [AltRoutingTel2IPMode]
None [0] = Alternative routing is not used. Conn [1] = Alternative routing is performed if ping to initial destination failed. QoS [2] = Alternative routing is performed if poor quality of service was detected. Both [3] = Alternative routing is performed if, either ping to initial destination failed, or poor quality of service was detected, or DNS host name is not resolved (default).
Note: QoS (Quality of Service) is quantified according to delay and packet loss, calculated according to previous calls. Qos statistics are reset if no new data is received for two minutes. For information on the Alternative Routing feature refer to
8.3 on page 152.
Max Allowed Packet Loss for Alt Routing [%]
[IPConnQoSMaxAllowedPL]
Packet loss percentage at which the IP connection is considered a failure. The range is 1% to 20%. The default value is 20%.
Max Allowed Delay for Alt Routing [msec]
[IPConnQoSMaxAllowedDel ay]
Transmission delay (in msec) at which the IP connection is considered a failure. The range is 100 to 1000. The default value is 250 msec.
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5.8.4.2 Tel to IP Routing Table
The Tel to IP Routing Table is used to route incoming Tel calls to IP addresses. This routing table associates a called / calling telephone number’s prefixes with a destination IP address or with an FQDN (Fully Qualified Domain Name). When a call is routed through the VoIP gateway (Proxy isn’t used), the called and calling numbers are compared to the list of prefixes on the IP Routing Table (up to 50 prefixes can be configured); Calls that match these prefixes are sent to the corresponding IP address. If the number dialed does not match these prefixes, the call is not made.
When using a Proxy server, you do not need to configure the Tel to IP Routing Table. However, if you want to use fallback routing when communication with Proxy servers is lost, or to use the ‘Filter Calls to IP’ and ‘IP Security’ features, or to obtain different SIP URI host names (per called number) or to assign IP profiles, you need to configure the IP Routing Table.
Note that for the Tel to IP Routing table to take precedence over a Proxy for routing calls, set the parameter ‘PreferRouteTable’ to 1. The gateway checks the 'Destination IP Address' field in the 'Tel to IP Routing' table for a match with the outgoing call. Only if a match is not found, a Proxy is used.
Possible uses for Tel to IP Routing can be as follows:
Can fallback to internal routing table if there is no communication with the Proxy servers.
Call Restriction – (when Proxy isn’t used), reject all outgoing TelIP calls that are
associated with the destination IP address: 0.0.0.0.
IP Security – When the IP Security feature is enabled (SecureCallFromIP = 1), the VoIP
gateway accepts only those IPTel calls with a source IP address identical to one of the IP addresses entered in the Tel to IP Routing Table.
Filter Calls to IP – When a Proxy is used, the gateway checks the TelIP routing table
before a telephone number is routed to the Proxy. If the number is not allowed (number isn’t listed or a Call Restriction routing rule was applied), the call is released.
Always Use Routing Table – When this feature is enabled (AlwaysUseRouteTable = 1), even
if a Proxy server is used, the SIP URI host name in the sent INVITE message is obtained from this table. Using this feature users are able to assign a different SIP URI host name for different called and/or calling numbers.
Assign Profiles to destination address (also when a Proxy is used).
Alternative Routing – (When Proxy isn’t used) an alternative IP destination for telephone
number prefixes is available. To associate an alternative IP address to called telephone number prefix, assign it with an additional entry (with a different IP address), or use an FQDN that resolves to two IP addresses. Call is sent to the alternative destination when one of the following occurs:
No ping to the initial destination is available, or when poor QoS (delay or packet loss,
calculated according to previous calls) is detected, or when a DNS host name is not resolved. For detailed information on Alternative Routing, refer to Section 8.3 on page
152.
When a release reason that is defined in the ‘Reasons for Alternative Tel to IP Routing’
table is received. For detailed information on the ‘Reasons for Alternative Routing Tables’ refer to Section 5.8.4.5 on page 81.
Alternative routing (using this table) is commonly implemented when there is no response to an Invite message (after Invite retransmissions). The gateway then issues an internal 408 ‘No Response’ implicit release reason. If this reason is included in the ‘Reasons for Alternative Routing’ table, the gateway immediately initiates a call to the redundant destination using the next matched entry in the ‘Tel to IP Routing’ table. Note that if a domain name in this table is resolved to two IP addresses, the timeout for Invite retransmissions can be reduced by using the parameter ‘Number of RTX Before Hotswap’.
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Tip: Tel to IP routing can be performed either before or after applying the number
manipulation rules. To control when number manipulation is done, set the Tel to IP Routing Mode parameter (described in Table 5-11).
To configure the Tel to IP Routing table, take these 6 steps:
1. Open the ‘Tel to IP Routing’ screen (Protocol Management menu > Routing Tables
submenu > Tel to IP Routing option); the ‘Tel to IP Routing’ screen is displayed (shown in
Figure
5-12).
2. In the ‘Tel to IP Routing Mode’ field, select the Tel to IP routing mode (refer to Table
5-11).
3. In the ‘Routing Index’ drop-down list, select the range of entries that you want to edit.
4. Configure the Tel to IP Routing table according to Table
5-11.
5. Click the Submit button to save your changes.
6. To save the changes so they are available after a power fail refer to Section
5.12 on page
139.
Figure
5-12: Tel to IP Routing Table Screen
Table
5-11: Tel to IP Routing Table
Parameter Description
Tel to IP Routing Mode [RouteModeTel2IP]
Route calls before manipulation [0] = TelIP calls are routed before the number manipulation rules are applied (default). Route calls after manipulation [1] = TelIP calls are routed after the number manipulation rules are applied. Note: Not applicable if Proxy routing is used.
Destination Phone Prefix Each entry in the Destination Phone Prefix fields represents a called telephone number
prefix. The prefix can be 1 to 19 digits long. An asterisk (*) represents all numbers.
Source Phone Prefix Each entry in the Source Phone Prefix fields represents a calling telephone number
prefix. The prefix can be 1 to 19 digits long. An asterisk (*) represents all numbers.
Any telephone number whose destination number matches the prefix defined in the ‘Destination Phone Prefix’ field and its source number matches the prefix defined in the adjacent ‘Source Phone Prefix‘ field, is sent to the IP address entered in the ‘IP Address’ field. Note that Tel to IP routing can be performed according to a combination of source and destination phone prefixes, or using each independently.
Note 1: An additional entry of the same prefixes can be assigned to enable alternative routing. Note 2: For available notations that represent multiple numbers refer to Section 5.8.3.1 on page 71.
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Table 5-11: Tel to IP Routing Table
Parameter Description
Destination IP Address In each of the IP Address fields, enter the IP address that is assigned to these prefixes.
Domain names, such as domain.com, can be used instead of IP addresses. To discard outgoing IP calls, enter 0.0.0.0 in this field. Note: When using domain names, you must enter a DNS server IP address, or alternatively define these names in the ‘Internal DNS Table’.
Profile ID Enter the number of the IP profile that is assigned to the destination IP address defined in
the ‘Destination IP Address’ field.
Status A read only field representing the quality of service of the destination IP address.
N/A = Alternative Routing feature is disabled. OK = IP route is available Ping Error = No ping to IP destination, route is not available QoS Low = Bad QoS of IP destination, route is not available DNS Error = No DNS resolution (only when domain name is used instead of an IP address).
Parameter Name in ini File Parameter Format Prefix
Prefix = <Destination Phone Prefix>,<IP Address>,<Source Phone Prefix>,<Profile ID>
For example: Prefix = 20,10.2.10.2,202,1 Prefix = 10[340-451]xxx#,10.2.10.6,*,1 Prefix = *,gateway.domain.com,* Note 1: <destination / source phone prefix> can be single number or a range of numbers. For available notations refer to Section 5.8.3.1 on page 71.
Note 2: This parameter can appear up to 50 times. Note 3: Parameters can be skipped by using the sign "$$", for example:
Prefix = $$,10.2.10.2,202,1
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5.8.4.3 IP to Hunt Group Routing
The IP to Hunt Group Routing Table is used to route incoming IP calls to groups of channels called hunt groups. Calls are assigned to hunt groups according to any combination of the following three options (or using each independently):
Destination phone prefix
Source phone prefix
Source IP address
The call is then sent to the VoIP gateway channels assigned to that hunt group. The specific channel, within a hunt group, that is assigned to accept the call is determined according to the hunt group’s channel selection mode which is defined in the Hunt Group Settings table (Section
5.8.7 on page 91) or according to the global parameter ‘ChannelSelectMode’ (refer to Table 5-5
on page 59). Hunt groups can be used on both FXO and FXS gateways; however, usually they are used with FXO gateways.
Note: When a release reason that is defined in the ‘Reasons for Alternative IP to Tel Routing’ table is received for a specific IPTel call, an alternative hunt group for that call is available. To associate an alternative hunt group to an incoming IP call, assign it with an additional entry in the ‘IP to Hunt Group Routing’ table (repeat the same routing rules with a different hunt group ID).
For detailed information on the ‘Reasons for Alternative Routing Tables’ refer to Section
5.8.4.5
on page 81.
To use hunt groups you must also do the following.
You must assign a hunt group ID to the VoIP gateway channels on the Endpoint Phone
Number screen. For information on how to assign a hunt group ID to a channel, refer to Section
5.8.6 on page 89.
You can configure the Hunt Group Settings table to determine the method in which new calls
are assigned to channels within the hunt groups (a different method for each hunt group can be configured). For information on how to enable this option, refer to Section
5.8.7 on page
91. If a Channel Select Mode for a specific hunt group isn’t specified, then the global
"Channel Select Mode" parameter (defined in ‘General Parameters’ screen under ‘Advanced Parameters’) applies.
To configure the IP to Hunt Group Routing table, take these 6 steps:
1. Open the ‘IP to Hunt Group Routing’ screen (Protocol Management menu > Routing
Tables submenu > IP to Hunt Group Routing option); the ‘IP to Hunt Group Routing’ table
screen is displayed (shown in Figure
5-13).
Figure
5-13: IP to Hunt Group Routing Table Screen
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2. In the ‘IP to Tel Routing Mode’ field, select the IP to Tel routing mode (refer to Table 5-12).
3. In the ‘Routing Index’ drop-down list, select the range of entries that you want to edit (up to
24 entries can be configured).
4. Configure the IP to Hunt Group Routing table according to Table
5-12.
5. Click the Submit button to save your changes.
6. To save the changes so they are available after a power fail refer to Section
5.12 on page
139.
Table 5-12: IP to Hunt Group Routing Table
Parameter Description
IP to Tel Routing Mode [RouteModeIP2Tel]
Route calls before manipulation [0] = IPTel calls are routed before the number manipulation rules are applied (default). Route calls after manipulation [1] = IPTel calls are routed after the number manipulation rules are applied.
Destination Phone Prefix Each entry in the Destination Phone Prefix fields represents a called telephone number
prefix. The prefix can be 1 to 49 digits long. An asterisk (*) represents all numbers.
Source Phone Prefix Each entry in the Source Phone Prefix fields represents a calling telephone number
prefix. The prefix can be 1 to 49 digits long. An asterisk (*) represents all numbers.
Source IP Address Each entry in the Source IP Address fields represents the source IP address of an
IPTel call (obtained from the Contact header in the Invite message). Note: The source IP address can include the “x” wildcard to represent single
digits. For
example: 10.8.8.xx represents all the addresses between 10.8.8.10 to 10.8.8.99.
Any SIP incoming call whose destination number matches the prefix defined in the ‘Destination Phone Prefix’ field and its source number matches the prefix defined in the adjacent ‘Source Phone Prefix‘ field and its source IP address matches the address defined in the ‘Source IP Address’ field, is assigned to the hunt group entered in the field to the right of these fields. Note that IP to hunt group routing can be performed according to any combination of source / destination phone prefixes and source IP address, or using each independently. Note: For available notations that represent multiple numbers (used in the prefix columns), refer to Section
5.8.3.1 on
page 71. Hunt Group ID In each of the Hunt Group ID fields, enter the hunt group ID to which calls that match
these prefixes are assigned.
Profile ID Enter the number of the IP profile that is assigned to the routing rule.
Parameter Name in ini File
Parameter Format
PSTNPrefix
PSTNPrefix = a,b,c,d,e
a = Destination Number Prefix b = Hunt Group ID c = Source Number Prefix d = Source IP address (obtained from the Contact header in the Invite message) e = IP Profile ID
Selection of hunt groups (for IP to Tel calls) is according to destination number, source number and source IP address.
Note 1: To support the ‘in call alternative routing’ feature, Users can use two entries that support the same call, but assigned it with a different hunt groups. The second entree functions as an alternative selection if the first rule fails as a result of one of the release reasons listed in the AltRouteCauseIP2Tel table.
Note 2: An optional IP ProfileID (1 to 5) can be applied to each routing rule. Note 3: The Source IP Address can include the “x” wildcard to represent single
digits.
For example: 10.8.8.xx represents all IP addresses between 10.8.8.10 to 10.8.8.99. Note 4: For available notations that represent multiple numbers refer to Section 5.8.3.1 on page 71. Note 5: This parameter can appear up to 24 times.
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5.8.4.4 Internal DNS Table
The internal DNS table, similar to a DNS resolution, translates hostnames into IP addresses. This table is used when hostname translation is required (e.g., ‘Tel to IP Routing’ table, etc.). Two different IP addresses can be assigned to the same hostname. If the hostname isn’t found in this table, the gateway communicates with an external DNS server.
Assigning two IP addresses to hostname can be used for alternative routing (using the ‘Tel to IP Routing’ table).
To configure the internal DNS table, take these 7 steps:
1. Open the ‘Internal DNS Table’ screen (Protocol Management menu > Routing Tables
submenu > Internal DNS Table option); the ‘Internal DNS Table’ screen is displayed.
Figure
5-14: Internal DNS Table Screen
2. In the ‘DNS Name’ field, enter the hostname to be translated. You can enter a string up to 31
characters long.
3. In the ‘First IP Address’ field, enter the first IP address that the hostname is translated to.
4. In the ‘Second IP Address’ field, enter the second IP address that the hostname is translated
to.
5. Repeat steps 2 to 4, for each Internal DNS Table entry.
6. Click the Submit button to save your changes.
7. To save the changes so they are available after a power fail refer to Section
5.12 on page
139.
Table 5-13: Internal DNS ini File Parameter
Parameter Name in ini File Parameter Format
DNS2IP
DNS2IP = <Hostname>, <first IP address>, <second IP address>
For example: DNS2IP = Domainname.com, 10.8.21.4, 10.13.2.95
Note: This parameter can appear up to 10 times.
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5.8.4.5 Reasons for Alternative Routing
The Reasons for Alternative Routing screen includes two tables (TelIP and IPTel). Each table enables you to define up to 4 different release reasons. If a call is released as a result of one of these reasons, the gateway tries to find an alternative route to that call. The release reason for IPTel calls is provided in Q.931 notation. The release reason for TelIP calls is provided in SIP 4xx, 5xx and 6xx response codes. For TelIP calls an alternative IP address, for IPTel calls an alternative hunt group.
Refer to ‘Tel to IP Routing’ on page 75 for information on defining an alternative IP address. Refer to the ‘IP to Hunt Group Routing’ on page 78 for information on defining an alternative hunt group.
You can use this table for example:
For TelIP calls, when there is no response to an Invite message (after Invite retransmissions), and the gateway then issues an internal 408 ‘No Response’ implicit release reason.
For IPTel calls, when the destination is busy, and release reason #17 is issued or for other call releases that issue the default release reason (#3). Refer to ‘DefaultReleaseCause’ in Table
5-5.
Note: The reasons for alternative routing option for TelIP calls only applies when Proxy isn’t used.
To configure the reasons for alternative routing, take these 5 steps:
1. Open the ‘Reasons for Alternative Routing’ screen (Protocol Management menu > Routing
Tables submenu > Reasons for Alternative Routing option); the ‘Reasons for Alternative
Routing’ screen is displayed.
Figure
5-15: Reasons for Alternative Routing Screen
2. In the ‘IP to Tel Reasons’ table, from the drop-down list select up to 4 different call failure
reasons that invoke an alternative IP to Tel routing.
3. In the ‘Tel to IP Reasons’ table, from the drop-down list select up to 4 different call failure
reasons that invoke an alternative Tel to IP routing.
4. Click the Submit button to save your changes.
5. To save the changes so they are available after a power fail refer to Section
5.12 on page
139.
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Table 5-14: Reasons for Alternative Routing ini File Parameter
Parameter Name in ini File Parameter Format
AltRouteCauseTel2IP
AltRouteCauseTel2IP = <SIP Call failure reason from IP>
For example: AltRouteCauseTel2IP = 408 (Response timeout). AltRouteCauseTel2IP = 486 (User is busy).
Note: This parameter can appear up to 4 times.
AltRouteCauseIP2Tel
AltRouteCauseIP2Tel = <Call failure reason from Tel>
For example: AltRouteCauseIP2Tel = 3 (No route to destination). AltRouteCauseIP2Tel = 17 (Busy here).
Note: This parameter can appear up to 4 times.
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5.8.5 Configuring the Profile Definitions
Utilizing the Profiles feature, the MP-1xx provides high-level adaptation when connected to a variety of equipment (from both Tel and IP sides) and protocols, each of which require a different system behavior. Using Profiles, users can assign different Profiles (behavior) on a per-call basis, using the Tel to IP and IP to Hunt Group Routing tables, or associate different Profiles to the gateway’s endpoint(s). The Profiles contain parameters such as Coders, T.38 Relay, Voice and DTMF Gains, Silence Suppression, Echo Canceler, RTP DiffServ, Current Disconnect and more. The Profiles feature allows users to tune these parameters or turn them on or off, per source or destination routing and/or the specific gateway or its ports. For example, specific ports can be designated to have a profile which always uses G.711.
Each call can be associated with one or two Profiles, Tel Profile and (or) IP Profile. If both IP and Tel profiles apply to the same call, the coders and other common parameters of the preferred Profile (determined by the Preference option) are applied to that call. If the Preference of the Tel and IP Profiles is identical, the Tel Profile parameters are applied.
Note: The default values of the parameters in the Tel and IP Profiles are identical
to the Web/ini file parameter values. If a value of a parameter is changed in the Web/ini file, it is automatically updated in the Profiles correspondingly. After any parameter in the Profile is modified by the user, modifications to parameters in the Web/ini file no longer impact that Profile.
5.8.5.1 Coder Group Settings
Use the Coders Group Settings screen to define up to four different coder groups. These coder groups are used in the Tel and IP Profile Settings screens to assign different coders to Profiles.
To configure the coder group settings, take these 8 steps:
1. Open the ‘Coder Group Settings’ screen (Protocol Management menu > Profile
Definitions submenu > Coder Group Settings option); the ‘Coder Group Settings’ screen is
displayed.
Figure
5-16: Coder Group Settings Screen
2. In the ‘Coder Group ID’ drop-down list, select the coder group you want to edit (up to four
coder groups can be configured).
3. From the coder drop-down list, select the coder you want to use. For the full list of available
coders and their corresponding ptimes refer to Table
5-15.
Note: Each coder can appear only once.
4. From the drop-down list to the right of the coder list, select the size of the Voice Packet
(ptime) used with this coder in milliseconds. Selecting the size of the packet determines how
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many coder payloads are combined into one RTP (voice) packet.
Note 1: The ptime packetization period depends on the selected coder name. Note 2: If not specified, the ptime gets a default value. Note 3: The ptime specifies the maximum packetization time the gateway can receive.
5. Repeat steps 3 and 4 for the second to fifth coders (optional).
6. Repeat steps 2 to 5 for the second to forth coder groups (optional).
7. Click the Submit button to save your changes.
8. To save the changes so they are available after a power fail refer to Section
5.12 on page
139.
Note: In the current version, only the ptime of the first coder is sent in the SDP
section of the Invite message.
Table 5-15: ini File Coder Group Parameters
Parameter Description CoderName_ID
Coder list for Profiles (up to five coders in each group). The CoderName_ID parameter (ID from 1 to 4) provides groups of coders that can be associated with IP or Tel profiles.
You can select the following coders: g711Alaw64k – G.711 A-law. g711Ulaw64k – G.711 µ-law. g7231 – G.723.1 6.3 kbps (default). g7231r53 – G.723.1 5.3 kbps. g726 – G.726 ADPCM 32 kbps (Payload Type = 35). g729 G.729A.
The RTP packetization period (ptime, in msec) depends on the selected Coder name, and can have the following values:
g711 family – 10, 20, 30, 40, 50, 60, 80, 100, 120 (default=20). g729 – 10, 20, 30, 40, 50, 60, 80, 100, 120 (default=20). g723 family – 30, 60, 90, 120, 150 (default = 30). G.726 family – 10, 20, 30, 40, 50, 60, 80, 100, 120 (default=20)
Note: G.729B is supported if the coder G.729 is selected and ‘EnableSilenceCompression’ equals 1 or 2. ini file note 1: This parameter (CoderName_ID) can appear up to 20 times (five coders in four coder groups).
ini file note 2: The coder name is case-sensitive. ini file note 3: Enter in the format: Coder,ptime.
For example, the following three coders belong to coder group with ID=1: CoderName_1 = g711Alaw64k,20 CoderName_1 = g711Ulaw64k,40 CoderName_1 = g7231,90
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5.8.5.2 Tel Profile Settings
Use the Tel Profile Settings screen to define up to four different Tel Profiles. These Profiles are used in the ‘Endpoint Phone Number’ table to associate different Profiles to gateway’s endpoints, thereby applying different behavior to different MP-1xx ports.
To configure the Tel Profile settings, take these 9 steps:
1. Open the ‘Tel Profile Settings’ screen (Protocol Management menu > Profile Definitions
submenu > Tel Profile Settings option); the ‘Tel Profile Settings’ screen is displayed.
Figure
5-17: Tel Profile Settings Screen
2. In the ‘Profile ID’ drop-down list, select the Tel Profile you want to edit (up to four Tel Profiles
can be configured).
3. In the ‘Profile Name’ field, enter a name that enables you to identify the Profile intuitively and
easily.
4. In the ‘Profile Preference’ drop-down list, select the preference (1-10) of the current Profile.
The preference option is used to determine the priority of the Profile. If both IP and Tel profiles apply to the same call, the coders and other common parameters of the preferred Profile are applied to that call. If the Preference of the Tel and IP Profiles is identical, the Tel Profile parameters are applied. Note: If the coder lists of both IP and Tel Profiles apply to the same call, an intersection of the coders is performed (i.e., only common coders remain). The order of the coders is determined by the preference.
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5. Configure the Profile’s parameters according to your requirements. For detailed information
on each parameter refer to the description of the screen in which it is configured as an individual parameter.
6. In the ‘Coder Group’ drop-down list, select the coder group you want to assign to that Profile.
You can select the gateway’s default coders (refer to Section
5.8.1.3 on page 53) or one of
the coder groups you defined in the Coder Group Settings screen (refer to Section
5.8.5.1 on
page 83).
7. Repeat steps 2 to 6 for the second to fifth Tel Profiles (optional).
8. Click the Submit button to save your changes.
9. To save the changes so they are available after a power fail refer to Section
5.12 on page
139.
Table 5-16: ini File Tel Profile Settings
Parameter Description TelProfile_ID
TelProfile_<Profile ID> = <Profile Name>,<Preference>,<Coder Group ID>,<IsFaxUsed *>,<DJBufMinDelay *>, <DJBufOptFactor *>,<IPDiffServ *>,<ControlIPDiffServ*>,<DTMFVolume>,<InputGain>, <VoiceVolume>,<EnableReversePolarity>,<EnableCurrentDisconnect>, <EnableDigitDelivery>, <ECE>
For example: TelProfile_1 = FaxProfile,1,2,0,10,5,22,33,2,22,34,1,0,1,1 TelProfile_2 = ModemProfile,0,10,13,$$,$$,$$,$$,$$,0,$$,0,0,1,1
$$ = Not configured, the default value of the parameter is used. (*) = Common parameter used in both IP and Tel profiles.
Note: This parameter can appear up to 4 times.
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5.8.5.3 IP Profile Settings
Use the IP Profile Settings screen to define up to four different IP Profiles. These Profiles are used in the Tel to IP and IP to Hunt Group Routing tables to associate different Profiles to routing rules. IP Profiles can also be used when working with Proxy server (set ‘AlwaysUseRouteTable’ to 1).
To configure the IP Profile settings, take these 9 steps:
1. Open the ‘IP Profile Settings’ screen (Protocol Management menu > Profile Definitions
submenu > IP Profile Settings option); the ‘IP Profile Settings’ screen is displayed.
Figure
5-18: IP Profile Settings Screen
2. In the ‘Profile ID’ drop-down list, select the IP Profile you want to edit (up to four IP Profiles
can be configured).
3. In the ‘Profile Name’ field, enter a name that enables you to identify the Profile intuitively and
easily.
4. In the ‘Profile Preference’ drop-down list, select the preference (1-10) of the current Profile.
The preference option is used to determine the priority of the Profile. If both IP and Tel profiles apply to the same call, the coders and other common parameters of the preferred Profile are applied to that call. If the Preference of the Tel and IP Profiles is identical, the Tel Profile parameters are applied. Note: If the coder lists of both IP and Tel Profiles apply to the same call, an intersection of the coders is performed (i.e., only common coders remain). The order of the coders is determined by the preference.
5. Configure the Profile’s parameters according to your requirements. For detailed information
on each parameter refer to the description of the screen in which it is configured as an individual parameter.
6. In the ‘Coder Group’ drop-down list, select the coder group you want to assign to that Profile.
You can select the gateway’s default coders (refer to Section
5.8.1.3 on page 53) or one of
the coder groups you defined in the Coder Group Settings screen (refer to Section
5.8.5.1 on
page 83).
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7. Repeat steps 2 to 6 for the second to fifth IP Profiles (optional).
8. Click the Submit button to save your changes.
9. To save the changes so they are available after a power fail refer to Section
5.12 on page
139.
Table 5-17: ini File IP Profile Settings
Parameter Description IPProfile_ID
IPProfile_<Profile ID> = <Profile Name>,<Preference>,<Coder Group ID>,<IsFaxUsed *>,<DJBufMinDelay *>, <DJBufOptFactor *>,<IPDiffServ *>,<ControlIPDiffServ *>,<EnableSilenceCompression>, <RTPRedundancyDepth>
For example: IPProfile_1 = name1,2,1,0,10,13,15,44,1,1 IPProfile_2 = name2,$$,$$,$$,$,$$,$$,$$,$$,1
$$ = Not configured, the default value of the parameter is used. (*) = Common parameter used in both IP and Tel profiles.
Note: This parameter can appear up to 4 times.
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5.8.6 Configuring the Endpoint Phone Numbers
From the Endpoint Phone Numbers screen you can enable and assign telephone numbers, hunt groups (optional) and profiles to the VoIP gateway ports.
To configure the Endpoint Phone Numbers table, take these 4 steps:
1. Open the ‘Endpoint Phone Numbers Table’ screen (Protocol Management menu >
Endpoint Phone Numbers); the ‘Endpoint Phone Numbers Table’ screen is displayed.
Figure
5-19: Endpoint Phone Number Table Screen
2. Configure the Endpoint Phone Numbers according to Table
5-18. You must enter a number
in the Phone Number fields for each port that you want to use.
3. Click the Submit button to save your changes, or click the Register or Un-Register buttons
to save your changes and to register / unregister to a Proxy / Registrar.
4. To save the changes so they are available after a power fail refer to Section
5.12 on page
139.
Table
5-18: Endpoint Phone Numbers Table
Parameter Description
Channel(s) The numbers (1-8) in the Channel(s) fields represent the ports on the back of the VoIP
gateway. To enable a VoIP gateway channel, you must enter the port number on this screen. [n-m] represents a range of ports. For example, enter [1-4] to specify the ports from 1 to
4. Note: For FXO gateways, the number of defined endpoints must not exceed the number of connected physical lines.
Phone Number In each of the Phone Number fields, enter the telephone number that is assigned to that
channel. For a range of channels enter the first number in an ordered sequence. These numbers are also used for port allocation for IP to Tel calls, if the hunt group’s ‘Channel Select Mode’ is set to ‘By Phone Number’.
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Table 5-18: Endpoint Phone Numbers Table
Parameter Description
Hunt Group ID In each of the Hunt Group ID fields, enter the hunt group ID (1-99) assigned to the
channel(s). The same hunt group ID can be used for more than one channel and in more than one field.
The hunt group ID is an optional field that is used to define a group of common behavior channels that are used for routing IP to Tel calls. If an IP to Tel call is assigned to a hunt group, the call is routed to the channel or channels that correspond to the hunt group ID.
You can configure the Hunt Group Settings table to determine the method in which new calls are assigned to channels within the hunt groups (refer to Section
5.8.7 on page
91).
Note: If you enter a hunt group ID, you must configure the IP to Hunt Group Routing Table (assigns incoming IP calls to the appropriate hunt group). If you do not configure the IP to Hunt Group Routing Table, calls don’t complete. For information on how to configure this table refer to Section
5.8.4.3.
Profile ID Enter the number of the Tel profile that is assigned to the endpoints defined in the
‘Channel(s)’ field.
Parameter Name in ini File Parameter Format TrunkGroup_x
TrunkGroup_<Hunt Group ID> = <Starting channel> - <Ending channel>, <Phone Number>, <Tel Profile ID>
For example: TrunkGroup_1 = 1-4,100 TrunkGroup_2 = 5-8,200,1
Note 1: The numbering of channels starts with 1. Note 2: ‘Hunt Group ID’ can be set to any number in the range 1 to 99. Note 3: When ‘x’ (Hunt Group ID) is omitted, the functionality of the TrunkGroup
parameter is similar to the functionality of ChannelList and Channel2Phone parameters. Note 4: This parameter can appear up to 8 times for MP-108 Gateways and up to 24 times for MP-124 Gateways.
Note 5: An optional Tel ProfileID (1 to 5) can be applied to each group of channels.
ChannelList Note: TrunkGroup_x
parameter can be used instead.
List of phone numbers for MP-1xx channels a, b, c, d a = first channel. b = number of channels starting from “a”. c = the phone number of the first channel. d = Tel Profile ID assigned to the group of channels. For example: ChannelList = 0,8,101, defines phone numbers 101 to 108 for up to 8 MP­108 channels.
Note 1: The ini file can include up to 24 “ChannelList = “ entries. Note 2: The “ChannelList“ can be used instead of, or in addition to, Channel2Phone
parameter.
Channel2Phone
Phone number of channel. Its format: Channel2Phone = “<channel>, <number>” <channel> is 0...23. Example: “Channel2Phone = 0, 1002” Appears once for each channel: 8 times for MP-108, or 4 times for MP-104 and twice for MP-102. For 8-port and 24-port gateways it is suggested to use “TrunkGroup“ parameter, where in a single line, all gateway’s phone numbers can be defined. Note: When ‘Channel2Phone’ is used to define an endpoint, hunt group and profile can’t be assigned to that endpoint.
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5.8.7 Configuring the Hunt Group Settings
The Hunt Group Settings Table is used to determine the method in which new calls are assigned to channels within each hunt group. If such a rule doesn’t exist (for a specific hunt group), the global rule, defined by the Channel Select Mode parameter (Protocol Definition > General Parameters), applies.
To configure the Hunt Group Settings table, take these 7 steps:
1. Open the ‘Hunt Group Settings’ screen (Protocol Management menu > Hunt Group
Settings); the ‘Hunt Group Settings’ screen is displayed.
Figure
5-20: Hunt Group Settings screen
2. In the Routing Index drop-down list, select the range of entries that you want to edit (up to
24 entries can be configured).
3. In the Hunt Group ID field, enter the hunt group ID number.
4. In the Channel Select Mode drop-down list, select the Channel Select Mode that
determines the method in which new calls are assigned to channels within the hunt groups entered in the field to the right of this field. For information on available Channel Select
Modes refer to Table
5-19.
5. Repeat steps 4 and 5, for each defined hunt group.
6. Click the Submit button to save your changes.
7. To save the changes so they are available after a power fail refer to Section
5.12 on page
139.
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Table 5-19: Channel Select Modes
Mode Description
By phone number Select the gateway port according to the called number (refer to the note
below).
Cyclic Ascending Select the next available channel in ascending cycle order. Always select the
next higher channel number in the hunt group. When the gateway reaches the highest channel number in the hunt group, it selects the lowest channel number in the hunt group and then starts ascending again.
Ascending Select the lowest available channel. Always start at the lowest channel number
in the hunt group and if that channel is not available, select the next higher channel.
Cyclic Descending Select the next available channel in descending cycle order. Always select the
next lower channel number in the hunt group. When the gateway reaches the lowest channel number in the hunt group, it selects the highest channel number in the hunt group and then start descending again.
Descending Select the highest available channel. Always start at the highest channel
number in the hunt group and if that channel is not available, select the next lower channel.
Number + Cyclic Ascending First select the gateway port according to the called number (refer to the note
below). If the called number isn’t found, then select the next available channel in ascending cyclic order. Note that if the called number is found, but the port associated with this number is busy, the call is released.
Parameter Name in ini File
Parameter Format
TrunkGroupSettings
TrunkGroupSettings = <Hunt group ID>, <Channel select Mode>
For example: TrunkGroupSettings = 1,5
<Channel Select Mode> can accept the following values:
0 = By Phone Number
1 = Cyclic Ascending
2 = Ascending
3 = Cyclic Descending
4 = Descending
5 = Number + Cyclic Ascending
Note: This parameter can appear up to 24 times.
Note: The gateway’s port numbers are defined in the ‘Endpoint Phone Numbers’
table under the ‘Phone Number’ column. For detailed information on the ‘Endpoint Phone Numbers’ table refer to Section 5.8.6 on page 89).
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5.8.8 Configuring the Endpoint Settings
The Endpoint Settings screens enable you to configure port-specific parameters.
5.8.8.1 Authentication
The Authentication Table (normally used with FXS gateways) defines a username and password combination for authentication for each MP-1xx port.
The ‘Authentication Mode’ parameter (described in Table
5-2) determines if authentication is
performed per port or for the entire gateway. If authentication is performed for the entire gateway, this table is ignored.
Note that if either the username or password field is omitted, the port’s phone number (defined in
Table 5-18) and global password (refer to the parameter ‘Password’ described in Table
5-2) are
used instead.
To configure the Authentication Table, take these 6 steps:
1. Set the ‘Authentication Mode’ parameter to ‘Authentication per Endpoint’.
2. Open the ‘Authentication’ screen (Protocol Management menu > Endpoint Settings >
Authentication option); the ‘Authentication’ screen is displayed.
Figure
5-21: Authentication Screen
3. In the ‘User Name’ and ‘Password’ fields for a port, enter the username and password
combination respectively.
4. Repeat step 4 for each port.
5. Click the Submit button to save your changes.
6. To save the changes so they are available after a power fail refer to Section
5.12 on page
139.
Table
5-20: Authentication ini File Parameter
Parameter Name in ini File Parameter Format
Authentication_x
Authentication_<Port Number> = <Username>,<Password>
For example: Authentication_0 = david,14325 Authentication_1 = Alex,18552
Note: Using the sign “$$” enables the User to omit either the username or the password. For instance, Authentication_5 = $$, 152. In this case, endpoint 5’s phone number is used instead of username.
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5.8.8.2 Automatic Dialing
Use the Automatic Dialing Table to define telephone numbers that are automatically dialed when a specific port is used.
To configure the Automatic Dialing table, take these 6 steps:
1. Open the ‘Automatic Dialing’ screen (Protocol Management menu > Endpoint Settings
submenu > Automatic Dialing option); the ‘Automatic Dialing’ screen is displayed.
Figure
5-22: Automatic Dialing Table Screen
2. In the ‘Destination Phone Number’ field for a port, enter the telephone number to dial.
3. In the ‘Auto Dial Status’ field, select one of the following:
Enable [1] – When a port is selected, when making a call, the number in the Destination
Phone Number field is automatically dialed if phone is offhooked (for FXS gateways) or ring signal is applied to port (FXO gateways).
Disable [0] – The automatic dialing option on the specific port is disabled (the number in
the Destination Phone Number field is ignored).
Hotline [2] – When a phone is offhooked and no digit is pressed for
‘HotLineDialToneDuration’, the number in the Destination Phone Number field is automatically dialed (applies to FXS and FXO gateways).
4. Repeat step 3 for each port you want to use for Automatic Dialing.
5. Click the Submit button to save your changes.
6. To save the changes so they are available after a power fail refer to Section
5.12 on page
139.
Note 1: After a ring signal is detected, on an ‘Enabled’ FXO port, the gateway
initiates a call to the destination number without seizing the line. The line is seized only after the call is answered.
Note 2: After a ring signal is detected on a ‘Disabled’ or ‘Hotline’ FXO port, the
gateway seizes the line.
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Table 5-21: Automatic Dialing ini File Parameter
Parameter Name in ini File Parameter Format
TargetOfChannelX
TargetOfChannel<Port> = <Phone>,<Mode>
For example: TargetOfChannel0 = 1001,1 TargetOfChannel3 = 911,2
Note 1: The numbering of channels starts with 0. Note 2: Define this parameter for each gateway port you want to use for
Automatic Dialing. Note 3: This parameter can appear up to 8 times for MP-108 gateways and up to 24 times for MP-124 gateways.
5.8.8.3 Caller ID
Use the Caller Display Information screen to send (to IP) Caller ID information when a call is made using the VoIP gateway (relevant to both FXS and FXO). The person receiving the call can use this information for caller identification. The information on this table is sent in an INVITE message in the ‘From’ header. For information on Caller ID restriction according to destination /
source prefixes refer to Section
5.8.3 on page 68.
Note: If Caller ID name is detected on an FXO line (EnableCallerID = 1), it is used
instead of the Caller ID name defined in this table (FXO gateways only).
To configure the Caller ID table, take these 6 steps:
1. Open the ‘Caller Display Information’ screen (Protocol Management menu > Endpoint
Settings submenu > Caller ID option); the ‘Caller Display Information’ screen is displayed.
Figure
5-23: Caller Display Information Screen
2. In the Caller ID/Name field, enter the Caller ID string. The Caller ID string can contain up to
18 characters. Note that when the FXS gateway receives “Private” or “Anonymous” strings in the ‘From’ header, it doesn’t send the calling name or number to the Caller ID display.
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3. In the ‘Presentation’ field, select ‘Allowed’ [0] to send the string in the Caller ID/Name field
when a (TelIP) call is made using this VoIP gateway port. Select ‘Restricted’ [1] if you don’t want to send this string. Note that when ‘Presentation’ is set to ‘Restricted’, the parameter ‘Asserted Identity Mode’ must be set to ‘P-Asserted’. Note: The value of the ‘Presentation’ field can (optionally) be overridden by configuring the ‘Presentation’ parameter in the ‘Source Number Manipulation’ table. To maintain backward compatibility, when the strings “Private” or “Anonymous” are set in the Caller ID/Name field, the Caller ID is restricted and the value in the Presentation field is ignored.
4. Repeat steps 2 and 3 for each VoIP gateway port.
5. Click the Submit button to save your changes.
6. To save the changes so they are available after a power fail refer to Section
5.12 on page
139.
Table
5-22: Caller ID ini File Parameter
Parameter Name in ini File Parameter Format
CallerDisplayInfoX
CallerDisplayInfo<channel> = <Caller ID string>,<Restriction>
0 = Not restricted (default). 1 = Restricted.
For example: CallerDisplayInfo0 = Susan C.,0 CallerDisplayInfo2 = Mark M.,1
Note 1: The numbering of channels starts with 0. Note 2: This parameter can appear up to eight times for MP-108, and up
to 24 times for MP-124.
5.8.8.4 Call Forward
The VoIP gateway allows you to forward incoming IPTel calls (using 302 response) based on the VoIP gateway port to which the call is routed (applicable only to FXS gateways).
The Call Forwarding Table is applicable only if the Call Forward feature is enabled. To enable Call Forward set ‘Enable Call Forward’ to ‘Enable’ in the ‘Supplementary Services’ screen, or
‘EnableForward=1’ in the ini file (refer to Table
5-6).
To configure the Call Forward table, take these 4 steps:
1. Open the ‘Call Forward Table’ screen (Protocol Management menu > Endpoint Settings
submenu > Call Forward option); the ‘Call Forward Table’ screen is displayed.
Figure
5-24: Call Forwarding Table Screen
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2. Configure the Call Forward parameters for each port according to the table below.
3. Click the Submit button to save your changes.
4. To save the changes so they are available after a power fail refer to Section
5.12 on page
139.
Table
5-23: Call Forward Table
Parameter Description
Forward Type
Not in use [0] = Don’t forward incoming calls (default). On Busy [1] = Forward incoming calls when the gateway port is busy. Immediate [2] = Forward any incoming call to the Phone number specified. No reply [3] = Forward incoming calls that are not answered with the time specified in the ‘Time for No Reply Forward’ field. On busy or No reply [4] = Forward incoming calls when the port is busy or when calls are not answered after a configurable period of time. Do Not Disturb [5] = Immediately reject incoming calls.
Forward to Phone Number Enter the telephone number or URL (number@IP address) to which the call is
forwarded. Note: If this field only contains telephone number and Proxy isn’t used, the ‘forward to’ phone number must be specified in the ‘Tel to IP Routing’ table of the forwarding gateway.
Time for No Reply Forward
If you have set the Forward Type for this port to no reply, enter the number of seconds the VoIP gateway waits before forwarding the call to the phone number specified.
Parameter Name in ini File Parameter Format FwdInfo_x
FwdInfo_<Gateway Port Number (0 to 23)> = <Forward Type>, <Forwarded SIP User Identification>, <Timeout (in seconds) for No Reply>
For example: FwdInfo_0 = 1,1001 FwdInfo_1 = 1,2003@10.5.1.1 FwdInfo_2 = 3,2005,30
Note 1: The numbering of gateway ports starts with 0. Note 2: This parameter can appear up to 24 times for MP-124.
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5.8.8.5 Caller ID Permissions
The Caller ID Permission table is used to enable or disable (per port) the Caller ID generation (for FXS gateways) and detection (for FXO gateways). If a port isn’t configured, its Caller ID generation / detection are determined according to the global parameter ‘EnableCallerID’
(described in Table
5-6).
To configure the Caller ID Permission Table, take these 5 steps:
1. Open the ‘Caller ID Permission’ screen (Protocol Management menu > Endpoint Setting s
> Caller ID Permission option); the ‘Caller ID Permission ’ screen is displayed.
Figure
5-25: MP-1xx FXS Caller ID Permission Screen
2. In the ‘Caller ID’ field, select one of the following:
Enable – Enables Caller ID generation (FXS) or detection (FXO) for the specific port. Disable – Caller ID generation (FXS) or detection (FXO) for the specific port is disabled. Empty – Caller ID generation (FXS) or detection (FXO) for the specific port is determined
according to the parameter ‘EnableCallerID’ (described in Table 5-6).
3. Repeat step 2 for each port.
4. Click the Submit button to save your changes.
5. To save the changes so they are available after a power fail refer to Section
5.12 on page
139.
Table
5-24: Authentication ini File Parameter
Parameter Name in ini File Parameter Format
EnableCallerID_X
EnableCallerID_<Port> = <Caller ID>
Caller ID: 0 = Disabled (default). 1 = Enabled. If not configured, use the global parameter ‘EnableCallerID’.
Note 1: The numbering of ports starts with 0. Note 2: This parameter can appear up to eight times for MP-108, and up
to 24 times for MP-124.
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Version 4.4 99 March 2005
5.8.9 Configuring the FXO Parameters
Use this screen to configure the gateway’s specific FXO parameters.
To configure the FXO parameters, take these 4 steps:
1. Open the ‘FXO Settings’ screen (Protocol Management menu > FXO); the ‘FXO Settings’
screen is displayed.
Figure
5-26: FXO Settings Screen
2. Configure the FXO parameters according to Table
5-25.
3. Click the Submit button to save your changes.
4. To save the changes so they are available after a power fail refer to Section
5.12 on page
139.
Table
5-25: FXO Parameters (continues on pages 99 to 100)
Parameter Description
Dialing Mode [IsTwoStageDial]
One Stage [0] = One-stage dialing. Two Stage [1] = Two-stage dialing (default).
Used for IPMP-10x/FXO gateways calls. If two-stage dialing is enabled, then the FXO gateway seizes one of the PSTN/PBX lines without performing any dial, the remote User is connected over IP to PSTN/PBX, and all further signaling (dialing and Call Progress Tones) is performed directly with the PBX without the gateway’s intervention.
If one-stage dialing is enabled, then the FXO gateway seizes one of the available lines (according to Channel Select Mode parameter), and dials the destination phone number received in INVITE message. Use the ‘Waiting For Dial Tone’ parameter to specify whether the dialing should come after detection of dial tone, or immediately after seizing of the line.
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Table 5-25: FXO Parameters (continues on pages 99 to 100)
Parameter Description
Waiting For Dial Tone [IsWaitForDialTone]
No [0] = Don’t wait for dial tone. Yes [1] = Wait for dial tone (default).
Used for IPMP-1xx/FXO gateways, when ‘One Stage Dialing’ is enabled. If “wait for dial tone” is enabled, the FXO gateway dials the phone number (to the PSTN/PBX line) only after it detects a dial tone.
Note 1: The correct dial tone parameters should be configured in the Call Progress Tones file. Note 2: It can take the gateway 1 to 3 seconds to detect a dial tone (according to the dial tone configuration in the Call Progress Tones file).
If ‘Waiting For Dial Tone‘ is disabled, the FXO gateway immediately dials the phone number after seizing the PSTN/PBX line, without ‘listening’ to dial tone.
Time to Wait before Dialing [msec] [FXOWaitForDialTime]
Delay (in milliseconds) between the time the line is seized and dialing is begun. The default is 1000 msec.
Note: Applicable only to MP-10x/FXO for single stage dialing, when waiting for dial tone is disabled.
Ring Detection Timeout [sec]
[FXOBetweenRingTime]
Note: Applicable only to MP-10x/FXO gateways for TelIP calls.
The Ring Detection timeout is used differently for normal and for automatic dialing. If automatic dialing is not used, and if Caller ID is enabled, then the FXO gateway seizes the line after detection of the second ring signal (allowing detection of caller ID, sent between the first and the second rings). If the second ring signal doesn’t arrive for “Ring Detection Timeout” the gateway doesn’t initiate a call to IP. When automatic dialing is used, the FXO gateway initiates a call to IP when ringing signal is detected. The FXO line is seized only if the remote IP party answers the call. If the remote party doesn’t answer the call and the ringing signal stops for “Ring Detection Timeout”, the FXO gateway Releases the IP call. Usually set to a value between 5 to 8. The default is 8 seconds.
Reorder Tone Duration [sec] [TimeForReorderTone]
Busy or Reorder tone duration (seconds) the FXO gateway plays before releasing the line. The valid range is 0 to 100. The default is 10 seconds. Usually, after playing a Reorder / Busy tone for the specified duration, the FXS gateway, starts playing an Offhook Warning tone.
Note 1: Selection of Busy or Reorder tone is performed according to the release cause received from IP. Note 2: Refer also to the parameter ‘CutThrough’ (described in Table 5-5).
Answer Supervision [EnableVoiceDetection]
Yes [1] = FXO gateway sends 200 OK (to INVITE) message when speech/fax/modem is detected. No [0] = 200 OK is sent immediately after the FXO gateway finishes dialing (default).
Note 1: To activate this feature set “DSPVersionTemplateNumber” parameter to 2 or 3. Usually this feature is used only with early media establish voice path before the call is answered. Note 2: This feature is applicable only to ‘One Stage’ dialing.
Rings before Detecting Caller ID
[RingsBeforeCallerID]
Sets the number of rings before the gateway starts detection of Caller ID (FXO only). 0 [0]= Before first ring. 1 [1]= After first ring (default). 2 [2]= After second ring.
SendMetering2IP
[Send Metering Message to IP]
No [0] = Disabled (default). Yes [1] = FXO gateways send a metering tone Info message to IP on detection of 12/16 kHz metering pulse. FXS gateways generate the 12/16 kHz metering tone on reception of a metering message. Note 1: Suitable (12 kHz or 16 kHz) coeff file must be used for both FXS and FXO gateways. The ‘MeteringType’ parameter must be defined in both FXS/FXO gateways. Note 2: The proprietary metering tone Info message is shown in Section
8.8 on page
159.
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