Dgw v2.0 Application Software Configuration Guide

Dgw v2.0 Application Software Configuration Guide
Discover the Power of 5
Software Configuration Guide
Dgw v2.0 Application
Document Revision 51
September 26,2016
Media5 Corporation
Software Configuration Guide
Dgw v2.0 Application Software Configuration Guide
© 2016, Media5 Corporation
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All other trademarks and registered trademarks are the property of their respective owners.
Third-Party Software Copyright Information
The Dgw v2.0 Application firmware aggregates some third-party software modules (open source and
commercial) that are distributed to you in accordance with their respective licenses. Refer to the Third
Party Software Copyright Information addendum available on the Mediatrix Download Portal, which
lists the third-party software modules along with any copyright and license information.
Software Configuration Guide
Contents
Preface
About this Manual ............................................................................................................ xix
Intended Audience..................................................................................................................................... xix
Related Documentation ............................................................................................................................. xix
Document Conventions .............................................................................................................................. xx
Warning Definition ......................................................................................................................................................xx
Other Conventions .....................................................................................................................................................xx
SCN vs. PSTN............................................................................................................................................................xx
Supported Standards .................................................................................................................................................xx
Naming Conventions ................................................................................................................................. xxi
Chapter 1
Information Important to Know.......................................................................................... 1
Introduction ...................................................................................................................................................1
Bypass Feature (Mediatrix 4100 Series) ......................................................................................................1
Management Choices...................................................................................................................................2
RESET/DEFAULT Button .............................................................................................................................4
User Access..................................................................................................................................................4
Secure Password Policies ........................................................................................................................................... 5
Chapter 2
Command Line Interface (CLI) ........................................................................................... 7
Introduction ...................................................................................................................................................7
Configuring the CLI.......................................................................................................................................7
Partial Reset................................................................................................................................................................ 8
Accessing the CLI.........................................................................................................................................8
Accessing the CLI through the Serial Console Port (Mediatrix 3000 Series only) ...................................................... 8
Accessing the CLI via a Telnet Session ...................................................................................................................... 9
Accessing the CLI via a SSH Session....................................................................................................................... 10
Working in the CLI ......................................................................................................................................11
Chapter 3
Web Interface Configuration ............................................................................................ 13
Introduction .................................................................................................................................................13
Tls Version Settings .................................................................................................................................................. 15
....................................................................................................................................................................16
HTTP User-Agent Header Format............................................................................................................................. 16
Using the Web Interface .............................................................................................................................18
Submitting Changes ...................................................................................................................................19
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System Parameters
Chapter 4
Services ............................................................................................................................. 23
Services Table ............................................................................................................................................23
Graceful Restart of Services ..................................................................................................................................... 26
Restarting a Service via MIB ..................................................................................................................................... 26
Chapter 5
Hardware Parameters ....................................................................................................... 29
Chapter 6
Endpoints State Configuration ........................................................................................ 31
Unit Configuration .......................................................................................................................................31
Endpoints Configuration .............................................................................................................................31
Administration .............................................................................................................................................32
Unit Shutting Down Behaviour .................................................................................................................................. 33
Chapter 7
Syslog Configuration ........................................................................................................ 35
Syslog Daemon Configuration ....................................................................................................................35
Configuring PCM Capture ......................................................................................................................................... 38
Configuring the Syslog Daemon Application ............................................................................................................. 38
Chapter 8
Events Configuration ........................................................................................................ 39
Notification Events ......................................................................................................................................39
Deleting a Rule.......................................................................................................................................................... 41
Monitoring Parameters .............................................................................................................................................. 41
Chapter 9
Local Log ........................................................................................................................... 43
Local Log Status and Entries......................................................................................................................43
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Chapter 10
VM ....................................................................................................................................... 45
Network Parameters
Chapter 11
IPv4 vs. IPv6 ...................................................................................................................... 49
Introduction .................................................................................................................................................49
IPv4 vs. IPv6 Availability.............................................................................................................................49
IPv6 Scope Identifier...................................................................................................................................50
When Contacting the unit using its IPv6 link-local Address ...................................................................................... 50
When Configuring the Mediatrix unit to use an IPv6 link-local Address .................................................................... 50
Chapter 12
Host Parameters................................................................................................................ 53
General Configuration.................................................................................................................................53
Host Configuration ......................................................................................................................................53
Default Gateway Configuration...................................................................................................................55
DNS Configuration......................................................................................................................................56
SNTP Configuration....................................................................................................................................57
Time Configuration .....................................................................................................................................58
STD / DST ................................................................................................................................................................. 59
OFFSET .................................................................................................................................................................... 59
START / END ............................................................................................................................................................ 59
Example .................................................................................................................................................................... 59
Additional Parameters ................................................................................................................................60
Configuring DNS Records Randomization ................................................................................................................ 60
Configuring Pre-resolved Static FQDNs.................................................................................................................... 61
Updating the "sysname" or "syslocation"................................................................................................................... 61
Chapter 13
Interface Parameters......................................................................................................... 63
Reserving an IP Address ............................................................................................................................63
Link Connectivity Detection ........................................................................................................................63
Partial Reset ...............................................................................................................................................63
Interfaces Configuration..............................................................................................................................64
IPv6 Autoconfiguration Interfaces ............................................................................................................................. 67
Network Interface Priority .......................................................................................................................................... 68
Rescue Interface Configuration ..................................................................................................................68
PPPoE Configuration..................................................................................................................................69
PPP Negotiation ........................................................................................................................................................ 70
DHCP Client Identifier Presentation .......................................................................................................................... 70
LLDP Configuration ....................................................................................................................................71
Ethernet Link Configuration ........................................................................................................................72
EAP 802.1x Configuration ..........................................................................................................................73
DHCP Server Configuration .......................................................................................................................73
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DHCP Negotiation ..................................................................................................................................................... 74
Ethernet Connection Speed........................................................................................................................75
Speed and Duplex Detection Issues ......................................................................................................................... 76
Chapter 14
VLAN Parameters .............................................................................................................. 77
Chapter 15
Local QoS (Quality of Service) Configuration ................................................................ 79
Introduction .................................................................................................................................................79
Differentiated Services (DS) Field .............................................................................................................79
IEEE 802.1q................................................................................................................................................81
Specific Service Class Configuration..........................................................................................................81
Network Traffic Control Configuration.........................................................................................................82
Chapter 16
Local Firewall Configuration ............................................................................................ 85
Managing the Local Firewall .......................................................................................................................85
Partial Reset.............................................................................................................................................................. 85
Setting the Default Policy .......................................................................................................................................... 85
Creating/Editing a Firewall Rule ................................................................................................................................ 86
Moving a Firewall Rule .............................................................................................................................................. 88
Deleting a Firewall Rule ............................................................................................................................................ 89
Disabling the Local Firewall ........................................................................................................................89
Chapter 17
IP Routing Configuration.................................................................................................. 91
Managing IP Routing ..................................................................................................................................91
IPv4 Forwarding ........................................................................................................................................................ 91
Creating/Editing an IP Routing Rule.......................................................................................................................... 92
Moving an IP Routing Rule........................................................................................................................................ 93
Deleting an IP Routing Rule ...................................................................................................................................... 93
Static IPv4 Routes..................................................................................................................................................... 94
DHCPv4 Classless Static Route Option .................................................................................................................... 95
DHCPv4 User Class Route Option............................................................................................................................ 95
Network Configuration Examples ...............................................................................................................96
Forward Packets from the Lan1 Network to the Uplink Network with NAT ............................................................... 96
Configure Port Forwarding for a Web Server Located on the LAN ........................................................................... 96
Chapter 18
Network Firewall Configuration ....................................................................................... 99
Managing the Network Firewall ..................................................................................................................99
Setting the Default Policy .......................................................................................................................................... 99
Creating/Editing a Network Firewall Rule ................................................................................................................ 100
Moving a Network Firewall Rule .............................................................................................................................. 103
Deleting a Network Firewall Rule ............................................................................................................................ 103
Disabling the Network Firewall .................................................................................................................103
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Chapter 19
NAT Configuration .......................................................................................................... 105
Introduction ...............................................................................................................................................105
Partial Reset............................................................................................................................................................ 105
Creating/Editing a Source NAT Rule ........................................................................................................105
Creating/Editing a Destination NAT Rule..................................................................................................109
Moving a NAT Rule...................................................................................................................................112
Deleting a NAT Rule .................................................................................................................................112
Disabling the NAT.....................................................................................................................................112
Chapter 20
DHCP Server Settings..................................................................................................... 113
Introduction ...............................................................................................................................................113
Subnet Server ......................................................................................................................................................... 113
Leases..................................................................................................................................................................... 113
Configuration Parameters ....................................................................................................................................... 113
Default vs. Specific Configurations.......................................................................................................................... 114
DHCP Basic Configuration .......................................................................................................................115
Lease Time (Option 51) ............................................................................................................................115
Domain Name (Option 15)........................................................................................................................116
Default Gateway (Option 3) ......................................................................................................................116
DNS (Option 6) .........................................................................................................................................117
NTP (Option 42)........................................................................................................................................118
NBNS (Option 44).....................................................................................................................................119
DHCP Static Leases Configuration...........................................................................................................120
SBC Parameters
Chapter 21
SBC Configuration .......................................................................................................... 123
POTS Parameters
Chapter 22
POTS Configuration ........................................................................................................ 127
POTS Status.............................................................................................................................................127
Line Status .............................................................................................................................................................. 127
FXO Line Status ...................................................................................................................................................... 128
General POTS Configuration....................................................................................................................128
Caller ID Information ............................................................................................................................................... 130
FXS Configuration ....................................................................................................................................131
FXS Country Customization .....................................................................................................................133
Calling Party Name of the Caller ID ........................................................................................................................ 134
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FXS Bypass ..............................................................................................................................................135
FXS Emergency Call Override..................................................................................................................136
FXS Distinctive Ring .................................................................................................................................137
FXO Configuration ....................................................................................................................................139
FXO Dialing Configuration....................................................................................................................................... 139
FXO Answering Configuration................................................................................................................................. 140
FXO Incoming Call Configuration............................................................................................................................ 142
FXO Custom Basic Parameters .............................................................................................................................. 142
FXO Line Verification .............................................................................................................................................. 143
FXO Force End of Call ............................................................................................................................................ 144
ISDN Parameters
Chapter 23
ISDN Configuration ......................................................................................................... 149
Introduction ...............................................................................................................................................149
ISDN Reference Points ........................................................................................................................................... 149
Inband Tones Generation........................................................................................................................................ 150
Setting PRI Hardware Parameters ...........................................................................................................151
ISDN Auto-Configuration ..........................................................................................................................152
Preset .......................................................................................................................................................153
Partial Reset............................................................................................................................................................ 153
PRI ISDN Statistics...................................................................................................................................153
PRI Configuration .....................................................................................................................................155
InformationFollowing Operation .............................................................................................................................. 166
BRI Configuration .....................................................................................................................................167
Bypass Feature (Mediatrix 3404/3408/3734/3741/3742 Models)............................................................................ 177
Bypass Feature (Mediatrix 4402plus / 4404plus Models) ....................................................................................... 177
Interop Parameters Configuration.............................................................................................................178
Play Local Ringback when no Media Stream .......................................................................................................... 180
ISDN Timers Configuration .......................................................................................................................182
Services Configuration..............................................................................................................................183
R2 CAS Parameters
Chapter 24
R2 CAS Configuration..................................................................................................... 191
Introduction ...............................................................................................................................................191
Line Signals for the Digital Version of MFC/R2 ....................................................................................................... 191
Interregister Signals ................................................................................................................................................ 191
Selecting the R2 Signaling Protocol .........................................................................................................192
R2 Auto-Configuration ..............................................................................................................................193
Preset .......................................................................................................................................................193
Partial Reset............................................................................................................................................................ 194
R2 Channel Associated Signaling ............................................................................................................194
R2 Signaling Variants ...............................................................................................................................198
Override Default Country Settings........................................................................................................................... 199
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R2 Signaling Variants.............................................................................................................................................. 199
R2 Timers Variants ...................................................................................................................................202
Override Default Country Settings........................................................................................................................... 203
R2 Timers Variants.................................................................................................................................................. 204
R2 Digit Timers Variants...........................................................................................................................207
Override Default Country Settings........................................................................................................................... 207
R2 Digit Timers Variants ......................................................................................................................................... 208
R2 Link Timers Variants ...........................................................................................................................209
Override Default Country Settings........................................................................................................................... 209
R2 Link Timers Variants .......................................................................................................................................... 210
R2 Tones Variants ....................................................................................................................................210
Override Default Country Settings........................................................................................................................... 212
R2 Tones Forward Groups ...................................................................................................................................... 213
R2 Tones Backward Groups ................................................................................................................................... 214
PRI R2 CAS Statistics ..............................................................................................................................218
E&M CAS Parameters
Chapter 25
E&M CAS Configuration ................................................................................................. 223
Introduction ...............................................................................................................................................223
Line Signals for the Digital Version of E&M............................................................................................................. 223
Selecting the E&M Signalling Protocol .....................................................................................................223
E&M Auto-Configuration ...........................................................................................................................224
Preset .......................................................................................................................................................225
Partial Reset............................................................................................................................................................ 226
E&M Channel Associated Signaling .........................................................................................................226
E&M Signalling Variants ...........................................................................................................................229
Override Default Signaling Settings ........................................................................................................................ 229
E&M Signalling Variants.......................................................................................................................................... 230
E&M Timers Variants................................................................................................................................232
Override Default Signaling Settings ........................................................................................................................ 233
E&M Timers Variants .............................................................................................................................................. 233
E&M Digit Timers Variants........................................................................................................................234
Override Default Signaling Settings ........................................................................................................................ 235
E&M Digit Timers Variants ...................................................................................................................................... 235
E&M Link Timers Variants ........................................................................................................................236
Override Default Signalling Settings........................................................................................................................ 236
E&M Link Timers Variants....................................................................................................................................... 237
E&M Tones Variants.................................................................................................................................237
Override Default Signalling Settings........................................................................................................................ 237
E&M Tones Variants ............................................................................................................................................... 238
PRI E&M Statistics....................................................................................................................................239
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SIP Parameters
Chapter 26
SIP Gateways................................................................................................................... 243
SIP Gateways Configuration.....................................................................................................................243
Chapter 27
SIP Servers ...................................................................................................................... 247
Introduction ...............................................................................................................................................247
TLS Persistent Connections Status ..........................................................................................................248
SIP Servers Configuration ........................................................................................................................248
Multiple SIP Gateways..............................................................................................................................249
SIP Gateway Specific Registrar Servers................................................................................................................. 249
SIP Gateway Specific Messaging Servers .............................................................................................................. 250
SIP Gateway Specific Proxy Servers ...................................................................................................................... 251
Keep Alive ............................................................................................................................................................... 251
SIP Gateway Specific Keep Alive Destinations ....................................................................................................... 253
Outbound Proxy Loose Router Configuration...........................................................................................253
Chapter 28
SIP Registration............................................................................................................... 255
Endpoints Registration..............................................................................................................................255
Contact Domain....................................................................................................................................................... 256
Accept Language .................................................................................................................................................... 257
Unit Registration .......................................................................................................................................257
Registration Configuration ........................................................................................................................258
Number of Registrations ...........................................................................................................................259
Additional Registration Refresh Parameters.............................................................................................260
Default Registration Retry Time .............................................................................................................................. 260
Default vs. Specific Configurations.......................................................................................................................... 260
Registration Refresh................................................................................................................................................ 261
Registration Expiration ............................................................................................................................................ 261
Expiration Value in Registration .............................................................................................................................. 261
Gateway Specific Registration Retry Time.............................................................................................................. 262
Unregistered Endpoint Behaviour ........................................................................................................................... 262
Unregistered Unit Behaviour ................................................................................................................................... 263
Behaviour on Initial-Registration Reception ............................................................................................................ 264
Registration Delay Value......................................................................................................................................... 265
SIP User Agent Header ............................................................................................................................265
Chapter 29
SIP Authentication .......................................................................................................... 267
Authentication Configuration.....................................................................................................................267
Creating/Editing an Authentication Entry................................................................................................................. 268
Moving an Authentication Entry............................................................................................................................... 270
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Deleting an Authentication Entry ............................................................................................................................. 270
Chapter 30
SIP Transport Parameters .............................................................................................. 271
SIP Transport Type...................................................................................................................................271
Additional Transport Parameters ..............................................................................................................273
Transport TLS Cipher Suite Settings....................................................................................................................... 273
Transport Tls Version Settings ................................................................................................................................ 274
................................................................................................................................................................................ 275
UDP Source Port Behaviour.................................................................................................................................... 275
TLS Client Authentication........................................................................................................................................ 276
Force DNS NAPTR In TLS...................................................................................................................................... 276
SIP Failover Conditions ........................................................................................................................................... 277
Persistent Port Interval ............................................................................................................................................ 278
Chapter 31
Interop Parameters.......................................................................................................... 279
Behavior on T.38 INVITE Not Accepted ...................................................................................................279
SIP Interop................................................................................................................................................279
SDP Interop ..............................................................................................................................................282
TLS Interop ...............................................................................................................................................286
Misc Interop ..............................................................................................................................................287
Additional Interop Parameters ..................................................................................................................288
Call Waiting Private Number Criteria for SIP INFO ................................................................................................. 288
Max-Forwards Header............................................................................................................................................. 288
Direction Attributes in a Media Stream .................................................................................................................... 289
Local Ring Behaviour on Provisional Response ..................................................................................................... 292
Session ID and Session Version Number in the Origin Field of the SDP................................................................ 293
Register Home Domain Override ............................................................................................................................ 294
DNS SRV Record Lock ........................................................................................................................................... 294
Listening for Early RTP ........................................................................................................................................... 294
Resolve Route Header ............................................................................................................................................ 295
ACK Branch Matching ............................................................................................................................................. 296
Ignore Require Header............................................................................................................................................ 296
Reject Code for Unsupported SDP Offer ................................................................................................................ 297
SIP User-Agent Header Format .............................................................................................................................. 297
SIP INFO Without Content Answer ......................................................................................................................... 298
Keep Alive Option Format ....................................................................................................................................... 298
Unsupported Content-Type ..................................................................................................................................... 299
Chapter 32
Miscellaneous SIP Parameters ...................................................................................... 301
SIP Penalty Box........................................................................................................................................301
Penalty Box vs Transport Types ............................................................................................................................. 301
Penalty Box Configuration....................................................................................................................................... 302
Error Mapping ...........................................................................................................................................302
SIP to Cause Error Mapping ................................................................................................................................... 304
Cause to SIP Error Mapping ................................................................................................................................... 306
Additional Headers ...................................................................................................................................309
PRACK .....................................................................................................................................................310
Forked Provisional Responses Behaviour .............................................................................................................. 311
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Session Refresh .......................................................................................................................................311
Background Information .......................................................................................................................................... 312
SIP Gateway Configuration ......................................................................................................................313
SIP Blind Transfer Method........................................................................................................................314
Diversion Configuration ............................................................................................................................314
DNS Configuration....................................................................................................................................315
Event Handling Configuration ...................................................................................................................315
Messaging Subscription............................................................................................................................317
Advice of Charge Configuration................................................................................................................317
Additional DNS Parameters......................................................................................................................318
DNS Failure Concealment....................................................................................................................................... 318
Media Parameters
Chapter 33
Voice & Fax Codecs Configuration ............................................................................... 321
Codec Descriptions...................................................................................................................................321
G.711 A-Law and µ-Law.......................................................................................................................................... 321
G.723.1.................................................................................................................................................................... 322
G.726....................................................................................................................................................................... 322
G.729....................................................................................................................................................................... 322
Clear Mode.............................................................................................................................................................. 323
Clear Channel ......................................................................................................................................................... 323
X-CCD Clear Channel ............................................................................................................................................. 324
T.38 ......................................................................................................................................................................... 324
Codec Parameters....................................................................................................................................325
Codec vs. Bearer Capabilities Mapping....................................................................................................326
Generic Voice Activity Detection (VAD) ....................................................................................................328
G.711 Codec Parameters .........................................................................................................................328
G.723 Codec Parameters .........................................................................................................................330
G.726 Codecs Parameters .......................................................................................................................331
G.729 Codec Parameters .........................................................................................................................333
Clear Mode Codec Parameters ................................................................................................................334
Clear Channel Codec Parameters............................................................................................................335
X-CCD Clear Channel Codec Parameters ...............................................................................................337
Fax Parameters ........................................................................................................................................338
Clear Channel Fax .................................................................................................................................................. 339
T.38 Fax .................................................................................................................................................................. 339
T.38 Parameters Configuration ............................................................................................................................... 340
Data Codec Selection Procedure ............................................................................................................................ 342
Chapter 34
Security ............................................................................................................................ 345
Introduction ...............................................................................................................................................345
Security Parameters .................................................................................................................................345
Enforcing Symmetric RTP ....................................................................................................................................... 347
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Chapter 35
RTP Statistics Configuration.......................................................................................... 349
Statistics Displayed...................................................................................................................................349
Statistics Configuration .............................................................................................................................351
Channel Statistics .....................................................................................................................................352
Chapter 36
Miscellaneous Media Parameters .................................................................................. 355
Jitter Buffer Configuration .........................................................................................................................355
About Changing Jitter Buffer Values ....................................................................................................................... 357
Starting a Call in Voiceband Data Mode ................................................................................................................. 358
DTMF Transport Configuration .................................................................................................................358
DTMF Transport over the SIP Protocol ................................................................................................................... 360
DTMF Detection ...................................................................................................................................................... 360
Using the Payload Type Found in the Answer ........................................................................................................ 363
Quantity of initial packets sent to transmit a DTMF Out-of-Band using RTP .......................................................... 363
Machine Detection Configuration..............................................................................................................364
Base Ports Configuration..........................................................................................................................365
Telephony Parameters
Chapter 37
DTMF Maps Configuration.............................................................................................. 369
Introduction ...............................................................................................................................................369
Syntax.......................................................................................................................................................369
Special Characters .................................................................................................................................................. 370
How to Use a DTMF Map ........................................................................................................................................ 370
General DTMF Maps Parameters.............................................................................................................372
Configuring Timeouts per Endpoint......................................................................................................................... 373
Allowed DTMF Maps ................................................................................................................................373
Refused DTMF Maps................................................................................................................................375
Chapter 38
Call Forward Configuration ............................................................................................ 377
Call Forward On Busy...............................................................................................................................377
Configuring Call Forward on Busy via Handset....................................................................................................... 379
Call Forward On No Answer .....................................................................................................................380
Configuring Call Forward on Answer via Handset................................................................................................... 381
Call Forward Unconditional.......................................................................................................................382
Configuring Call Forward on Unconditional via Handset ......................................................................................... 383
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Chapter 39
Telephony Services Configuration ................................................................................ 385
General Configuration...............................................................................................................................385
Automatic Call ......................................................................................................................................................... 387
Call Completion ....................................................................................................................................................... 387
Call Transfer ............................................................................................................................................................ 391
Call Waiting ............................................................................................................................................................. 393
Conference.............................................................................................................................................................. 397
Delayed Hot Line..................................................................................................................................................... 401
Direct IP Address Call ............................................................................................................................................. 402
Call Hold.................................................................................................................................................................. 403
Second Call ............................................................................................................................................................. 404
Message Waiting Indicator .......................................................................................................................405
Visual Message Waiting Indicator Type .................................................................................................................. 407
Distinctive Call Waiting Tone ....................................................................................................................407
Call Statistics ............................................................................................................................................408
Default Outbound Priority Call Routing.....................................................................................................409
Chapter 40
Tone Customization Parameters Configuration........................................................... 411
Current Tone Definition.............................................................................................................................411
Tone Override...........................................................................................................................................412
Chapter 41
Music on Hold Parameters Configuration..................................................................... 415
MP3 File Download Server .......................................................................................................................415
Configuring the TFTP Server .................................................................................................................................. 415
Configuring the HTTP Server .................................................................................................................................. 415
Music on Hold Configuration.....................................................................................................................415
Chapter 42
Country Parameters Configuration ............................................................................... 419
Country Configuration...............................................................................................................................419
Additional Country Settings ......................................................................................................................421
Default vs. Specific Configurations.......................................................................................................................... 421
Input/Output User Gain ........................................................................................................................................... 421
Dialing Settings ....................................................................................................................................................... 422
Fax Calling Tone Detection ..................................................................................................................................... 424
Chapter 43
Call Detail Record............................................................................................................ 425
CDR (Call Detail Record)..........................................................................................................................425
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Call Router Parameters
Chapter 44
Call Router Configuration............................................................................................... 431
Introduction ...............................................................................................................................................431
Limitations ............................................................................................................................................................... 432
Regular Expressions ............................................................................................................................................... 432
Routing Type ........................................................................................................................................................... 434
Call Properties Parameters ..................................................................................................................................... 440
SIP/ISDN Call Default Values ................................................................................................................................. 447
Call Routing Status.................................................................................................................................................. 448
Routes ......................................................................................................................................................449
Creating/Editing a Route ......................................................................................................................................... 450
Moving a Route ....................................................................................................................................................... 454
Deleting a Route...................................................................................................................................................... 455
Mappings ..................................................................................................................................................455
Creating/Editing a Mapping Type ............................................................................................................................ 455
Creating/Editing a Mapping Expression .................................................................................................................. 457
Moving a Mapping Type or Expression Row........................................................................................................... 464
Deleting a Mapping Type or Expression Row ......................................................................................................... 464
Signalling Properties.................................................................................................................................465
Creating/Editing a Signalling Property..................................................................................................................... 465
Moving a Signalling Property Row .......................................................................................................................... 469
Deleting a Signalling Property Row......................................................................................................................... 469
SIP Headers Translations .........................................................................................................................469
Creating/Editing a SIP Headers Translation............................................................................................................ 469
Moving a SIP Headers Translation Row ................................................................................................................. 471
Deleting a SIP Headers Translation Row................................................................................................................ 471
Call Properties Translations......................................................................................................................472
Creating/Editing a Call Properties Translation ........................................................................................................ 472
Moving a Call Properties Translation Row .............................................................................................................. 474
Deleting a Call Properties Translation Row............................................................................................................. 474
Hunt Service .............................................................................................................................................475
Creating/Editing a Hunt ........................................................................................................................................... 475
Call Rejection (Drop) Causes .................................................................................................................................. 478
Moving a Hunt ......................................................................................................................................................... 482
Deleting a Hunt........................................................................................................................................................ 482
SIP Redirects............................................................................................................................................483
Creating/Editing a SIP Redirect............................................................................................................................... 483
Moving a SIP Redirect............................................................................................................................................. 484
Deleting a SIP Redirect ........................................................................................................................................... 484
Hairpinning................................................................................................................................................485
Configuration Examples............................................................................................................................485
Chapter 45
Auto-Routing Configuration........................................................................................... 487
Auto-Routing.............................................................................................................................................487
Endpoints Auto-Routing .......................................................................................................................................... 488
Manual Routing ....................................................................................................................................................... 490
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Management Parameters
Chapter 46
Configuration Script........................................................................................................ 495
Chapter 47
Configuration Backup/Restore ...................................................................................... 497
Chapter 48
Firmware Download ........................................................................................................ 499
Chapter 49
Certificates Management................................................................................................ 501
Introduction ...............................................................................................................................................501
HTTPS Transfer Cipher Suite Settings ................................................................................................................... 502
HTTPS Transfer Tls Version Settings ..................................................................................................................... 503
..................................................................................................................................................................504
Managing Certificates ...............................................................................................................................504
Certificate Authorities................................................................................................................................505
Certificate Upload through the Web Browser ...........................................................................................506
Transferring a Certificate via Configuration Script................................................................................................... 506
Host Certificate Associations ....................................................................................................................507
Chapter 50
SNMP Configuration ....................................................................................................... 509
Introduction ...............................................................................................................................................509
SNMP Configuration Section ....................................................................................................................509
Partial Reset............................................................................................................................................................ 513
SNMP Statistics ........................................................................................................................................514
Chapter 51
CWMP Configuration ...................................................................................................... 515
Introduction ...............................................................................................................................................515
Licence Key Activation of TR-069 ........................................................................................................................... 515
CWMP Configuration Section ...................................................................................................................516
General Configuration ............................................................................................................................................. 516
ACS Configuration................................................................................................................................................... 517
ACS Configuration Parameters ............................................................................................................................... 520
Periodic Inform Configuration.................................................................................................................................. 521
TR-069 Configuration .............................................................................................................................................. 523
TR-104 Configuration .............................................................................................................................................. 524
TR-106 Configuration .............................................................................................................................................. 525
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TR-111 Configuration .............................................................................................................................................. 526
Transport HTTPS Cipher Suite Settings ................................................................................................................. 528
HTTPS Transport Tls Version Settings ................................................................................................................... 530
..................................................................................................................................................................531
Transport Certificate Validation ............................................................................................................................... 531
Allow Unauthenticated UDP Connection Requests................................................................................................. 532
Parameter Type Validation...................................................................................................................................... 533
MAC Address Format .............................................................................................................................................. 533
ACS Access to the Local Log Table........................................................................................................................ 534
Supported TR-069 Methods and Parameters...........................................................................................535
Chapter 52
Access Control Configuration ....................................................................................... 537
Users ........................................................................................................................................................537
Partial Reset............................................................................................................................................................ 538
Services Access Control Type..................................................................................................................538
Partial Reset............................................................................................................................................................ 539
Radius Servers .........................................................................................................................................539
Access Rights Description ........................................................................................................................541
Chapter 53
File Manager .................................................................................................................... 543
File Manager.............................................................................................................................................543
Partial Reset............................................................................................................................................................ 545
Transfer Protocols ................................................................................................................................................... 546
Security Certificates ................................................................................................................................................ 546
HTTPS Transfer Settings ........................................................................................................................................ 546
HTTPS Transfer Cipher Suite Settings ................................................................................................................... 546
HTTPS Transfer Tls Version Settings ..................................................................................................................... 548
..................................................................................................................................................................549
Chapter 54
Miscellaneous.................................................................................................................. 551
Management Interface Configuration .......................................................................................................551
Activate Licence........................................................................................................................................552
Dgw v2.0 Application
xvii
Contents
xviii
Dgw v2.0 Application
Intended Audience
P
Software Configuration Guide
R E F A C E
P
About this Manual
Thank you for purchasing one or more Mediatrix units supporting the Dgw v2.0 application.
The Dgw v2.0 application runs on several Mediatrix devices. Provider-specific profiles ensure that the
Mediatrix unit is a genuine plug and play solution. It offers a low total cost of ownership as it reduces installation
and maintenance costs.
Intended Audience
This Software Configuration Guide is intended for the following users:

System administrators who are responsible for installing and configuring networking equipment
and who are familiar with the Mediatrix unit.

System administrators with a basic networking background and experience, but who might not
be familiar with the Mediatrix unit.



Operators.
Installers.
Maintenance technicians.
Related Documentation
In addition to this manual, the Mediatrix unit documentation, available at http://www.media5corp.com/
documentation set includes the following:

Hardware Installation Guides for each specific Mediatrix unit to install the hardware of your
specific Mediatrix unit.


Quick Starts for each specific Mediatrix unit to quickly setup and work with the Mediatrix unit.
The DGW v2.0 Reference Guide providing the complete description of:
•
Parameters, tables, commands and available Country Specifications
•
Error messages
•
Notification messages
•
Country Tone Definitions

Third Party Software Copyright Information listing the third-party software modules used in the
Mediatrix unit along with any copyright and license information.



Configuration Notes providing specific hands-on configuration scenarios.
Technical Bulletins providing information on a specific use of the DGW v2.0 software.
Release notes
Supported Standards providing information of the RFCs (Request for Comments) standards, Internet-Drafts,
or other standard documents the Dgw v2.0 application is based on. This document is available upon request
from our technical support team.Therefore, it is possible that some behaviour differs from the official
standards.For more information on and a list of RFCs and Internet-Drafts, refer to the IETF web site at http://
www.ietf.org.
Dgw v2.0 Application
xix
Preface - About this Manual
Document Conventions
Document Conventions
The following information provides an explanation of the symbols that appear on the Mediatrix unit and in the
documentation for the product.
Warning Definition
Warning: Means danger. You are in a situation that could cause bodily injury. Before you work on any
equipment, you must be aware of the hazards involved with electrical circuitry and be familiar with standard
practices for preventing accidents.
Where to find Translated Warning Definition
For safety and warning information, refer to the Mediatrix unit Hardware Installation Guide. These documents
describe the international agency compliance and safety information for the Mediatrix unit. They also include
a translation of the safety warning listed in the previous section.
Other Conventions
The following are other conventions you will encounter in this manual.
Caution: Indicates a potentially hazardous situation which, if not avoided, may result in minor or moderate
injury and/or damage to the equipment or property.
Note: Indicates important information about the current topic.
Standards Supported
Indicates which RFC, Draft or other standard document is supported for a
specific feature.
SCN vs. PSTN
In Media5’ and other vendor’s documentation, the terms SCN and PSTN are used. A SCN (Switched Circuit
Network) is a general term to designate a communication network in which any user may be connected to any
other user through the use of message, circuit, or packet switching and control devices. The Public Switched
Telephone Network (PSTN) or a Private Branch eXchange (PBX) are examples of SCNs.
Supported Standards
When available, this document lists the standards onto which features are based. These standards may be
RFCs (Request for Comments), Internet-Drafts, or other standard documents.
The Dgw v2.0 application’s implementations are based on the standards, so it’s possible that some behaviour
differs from the official standards.
For more information on and a list of RFCs and Internet-Drafts, refer to the IETF web site at http://www.ietf.org.
Refer to the Supported Standards document at http://www.media5corp.com/documentation
xx
Dgw v2.0 Application
Naming Conventions
Software Configuration Guide
Naming Conventions
When defining a name for a parameter, only ascii characters are authorised. This is valid when defining a
name for a parameter in a Web Page of the Management interface, but also for parameters accessed via the
CLI, the MIB, or a script.
For example, to be valid, the Service Name defined during PPPoE configuration must only contain ascii
characters. Special characters such as " " (space), """ (double quote), "“" (left double quote), "‘" (left single
quote), "#", "£", "¢", "¿", "¡", "«", "»" will cause the system to display a syntax error message.
Dgw v2.0 Application
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Preface - About this Manual
Naming Conventions
xxii
Dgw v2.0 Application
Introduction
C
Software Configuration Guide
H A P T E R
1
Information Important to Know
This chapter provides an overview of the Mediatrix devices supported by the Dgw v2.0 application:




Description of the Bypass feature for models that support it.
Description of the various ways to manage the Mediatrix unit.
How to use the DEFAULT/RESET button (partial reset and factory reset procedures).
How to configure user access to the Mediatrix unit.
Introduction
The Dgw v2.0 application runs on all Mediatrix devices. Provider-specific profiles ensure that the Mediatrix unit
is a genuine plug and play solution. It offers a low total cost of ownership as it reduces installation and
maintenance costs.
Moreover, the Mediatrix unit integrates features such as TLS, SRTP, and HTTPS designed to bring enhanced
security for network management, SIP signalling and media transmission aspects.
For the complete list of Mediatrix unit brochures and technical specifications go to the at http://
www.media5corp.com/documentation
Bypass Feature (Mediatrix 4100 Series)
During normal operation, the SCN line connected to the Bypass connector is switched out of the circuit through
commuting relays. The Bypass connector can be activated by two different conditions:


When power is removed from the Mediatrix unit.
When the IP network is down.
Note: The Mediatrix 4102S does not have the Bypass feature.
This is indicated by the In Use LED being steady ON (except when the power is removed). If one of these
conditions is met, a phone/fax used on FXS connector 1 (Mediatrix 4104/4108/4116) or analog line 1
(Mediatrix 4124) is directly connected to the SCN Bypass line. FXS connector 1 (Mediatrix 4104/4108/4116)
or analog line 1 (Mediatrix 4124) stays in Bypass connection until:


Dgw v2.0 Application
The error conditions have been cleared.
The device connected to it is on-hook and a delay has elapsed.
1
Chapter 1 - Information Important to Know
Management Choices
Management Choices
The Mediatrix unit offers various management options to configure the unit.
Figure 1: Management Interfaces
Local
Configuration
Management
Configuration Files
Mediatrix Unit
Manager
Network (UMN)
Provisioning Server
Command Line
Interface (CLI)
Mediatrix Unit
SNMP
Queries
Web GUI
User
External EMS
Table 1: Management Options
Management Choice
Web GUI
Description
The Mediatrix unit web interface offers
the following options:
•
•
Password-protected access
via basic HTTP
authentication, as described
in RFC 2617
User-friendly GUI
Refer to “Chapter 3 - Web Interface
Configuration” on page 13 for more
details.
SNMPv1/2/3
The Mediatrix unit SNMP feature offers
the following options:
•
Password-protected access
•
Remote management
•
Simultaneous management
Features
The Mediatrix unit web interface allows
you to configure the following
information:
•
Network attributes
•
SIP parameters
•
VoIP settings
•
Management settings such
as configuration scripts,
restore / backup, etc.
The Mediatrix unit SNMP feature allows
you to configure all the MIB services.
Refer to “Chapter 50 - SNMP
Configuration” on page 509 for more
details.
2
Dgw v2.0 Application
Management Choices
Software Configuration Guide
Table 1: Management Options (Continued)
Management Choice
Command Line
Interface (CLI)
Description
The Mediatrix unit uses a proprietary
CLI to configure all the unit’s
parameters.
Features
The Mediatrix unit CLI feature allows
you to configure all the MIB services.
Refer to “Chapter 2 - Command Line
Interface (CLI)” on page 7 for more
details.
Unit Manager
Network
The Unit Manager Network (UMN) is a
PC-Windows based element
management system designed to
facilitate the deployment, configuration
and provisioning of Mediatrix access
devices and gateways.
The UMN offers the following:
•
Auto-discovery
•
Group provisioning
•
SNMP access and remote
management.
The UMN enables the simple and
remote configuration and deployment of
numerous Mediatrix units.
Dgw v2.0 Application
3
Chapter 1 - Information Important to Know
RESET/DEFAULT Button
RESET/DEFAULT Button
The RESET/DEFAULT button allows you to:
Cancel an action that was started.
Revert to known factory settings if the Mediatrix unit refuses to work properly for any reason or
the connection to the network is lost.
Reconfigure a unit.
The Partial reset provides a way to contact the Mediatrix unit in a known and static state while keeping most of the
configuration unchanged. Refer to the Performing a Partial Reset Technical Bulletin http://www.media5corp.com/
documentation
The Factory reset reverts the Mediatrix unit back to its default factory settings. Refer to Performing a Factory Reset at
http://www.media5corp.com/documentation.
For the complete details on the RESET/DEFAULT button, refer to Using the RESET/DEFAULT Button Technical Bulletin
at http://www.media5corp.com/documentation.
User Access
This section describes configuration that is available only in the MIB parameters of the Mediatrix unit. You can
configure these parameters as follows:



by using a MIB browser
by using the CLI
by creating a configuration script containing the configuration variables
The following describes how to configure user access to the Mediatrix unit. The access information is available
for the SNMP and Web interface management methods.
Note: Currently, the user name cannot be modified. To access the unit via SNMPv1, you must use the user
name as being the “community name” and there must be no password for this user name.
 To configure the Mediatrix unit user access:
1.
In the aaaMIB, set the password associated with the user name in the usersPassword variable.
You can also use the following line in the CLI or a configuration script:
aaa.users.Password[UserName="User_Name"]="Value"
Only the “admin” and “public” user names are available for the moment.
2.
Set the user name that is used for scheduled tasks in the batchUser variable.
You can also use the following line in the CLI or a configuration script:
aaa.batchUser="Value"
For instance, if you are using an automatic configuration update everyday at midnight, the relevant
service will use the “batchUser” user to execute the request.
4
Dgw v2.0 Application
User Access
Software Configuration Guide
Secure Password Policies
It is possible to validate a password against some password policies to be considered as valid. These policies
may only be activated via customized profiles created by Media5. The available policies are:
Table 2: Secure Password Policies
Policy
Description
Minimum Length of User Password The minimum length the user password must have to be considered
as valid.
Upper and Lower Case Required
on User Password
Indicates if the user password is required to contain an upper and a
lower case characters to be considered as valid.
Here is an example of a valid password : 'Password' and examples of
invalid passwords : '1234', 'password', '1password', 1PASSWORD.
Numeral character Required on
User Password
Indicates if the user password is required to contain a numeral
character to be considered as valid.
Here is an example of a valid password : '1password2' and examples
of invalid passwords : 'password', 'Password'.
Special character Required on
User Password
Indicates if the user password is required to contain a special character
to be considered as valid.
Here is an example of a valid passwords : 'pass$word', 'pass_word#'
and examples of invalid passwords : 'password', 'Password', '1234',
'1Password'.
For more information on how to get a customized user profile, please refer to your Media5 representative.
Dgw v2.0 Application
5
Chapter 1 - Information Important to Know
6
User Access
Dgw v2.0 Application
Introduction
C
Software Configuration Guide
H A P T E R
2
Command Line Interface (CLI)
This chapter describes how to access the CLI environment in order to perform configuration tasks.





Introduction
Configuring the CLI
Accessing the CLI
•
Accessing the CLI through the Serial Console Port (Mediatrix 3000 Series only)
•
Accessing the CLI via a Telnet Session
•
Accessing the CLI via a SSH Session
Working in the CLI
•
Contexts
•
Exiting from the CLI
•
Command Completion
•
Macros
•
History
•
Service Restart
•
Configuring the Mediatrix unit with the CLI
List of Commands / Keywords
Introduction
You can configure the Mediatrix unit parameters through a proprietary Command Line Interface (CLI)
environment. It allows you to configure the unit parameters by Serial port (Mediatrix 3000 Series only), Telnet
or SSH.
The CLI uses the Media5 proprietary scripting language as described in the Scripting Language at http://
www.media5corp.com/documentation.
.
Configuring the CLI
You must configure the CLI access. This can be done via the MIB variables. Once you have access to the CLI,
you can also use it to configure the access.
 To configure the CLI access:
1.
In the cliMIB, set the inactivity expiration delay for exiting the CLI session in the
inactivityTimeOut variable.
If there is no activity during the delay defined, the CLI session is closed. This value is expressed in
minutes.
2.
Enable remote Telnet access if applicable by setting the EnableTelnet variable to Enable.
By default, Telnet is not enabled.
3.
Dgw v2.0 Application
Set the port on which the Telnet service should listen for incoming Telnet requests in the IpPort.
variable.
7
Chapter 2 - Command Line Interface (CLI)
Accessing the CLI
4.
Enable remote SSH access if applicable by setting the EnableSsh variable to Enable.
5.
Set the port on which the SSH service should listen for incoming SSH requests in the IpPort.
variable.
The configuration is loaded when it is started. It configures and starts Telnet and SSH according to the options
offered through the configuration variables. The configuration can be updated by the CLI service while running.
Partial Reset
When a partial reset is triggered, the CLI variables revert back to their default value.
Accessing the CLI
You can access the CLI through the console port of the Mediatrix unit (Mediatrix 3000 Series only) or through
a Telnet or SSH session.
Only one session at a time is allowed. These sections describe how to access the CLI:


“Accessing the CLI through the Serial Console Port (Mediatrix 3000 Series only)” on page 8

“Accessing the CLI via a SSH Session” on page 10
“Accessing the CLI via a Telnet Session” on page 9
•
“Opening a Telnet Session with the Unit Manager Network” on page 9
Which method you choose depends primarily on your preference and level of experience with one or all of the
options provided. None precludes using other configuration methods. Note that after performing a factory reset
or a firmware update, accessing the CLI may take up to one minute, even if the web and SNMP interfaces are
already accessible.
Note: When performing a partial reset, the root password is removed. For more details refer to Performing
a Partial Reset Technical Bulletin at http://www.media5corp.com/documentation.
Accessing the CLI through the Serial Console Port (Mediatrix 3000 Series only)
Serial console access requires a computer, serial terminal software, and null modem cable. Many networking
environments have serial terminals available – or a laptop configured to act as a serial terminal – setup for use
in configuration of network devices such as routers. In this case, it may be simplest to use the serial console
access.
 To access the CLI through the console port:
1.
Connect a null modem cable, or crossover serial cable, to the serial console port on the front panel
of the Mediatrix unit and to a serial port on your serial terminal computer.
2.
Using a terminal software, connect by using the following parameters:
3.
•
9600 baud
•
no parity
•
8 data bits
•
1 stop bit
•
no flow control
If the Mediatrix unit is not on, power it up. The startup display should appear on the screen. If the
Mediatrix unit is already started, you will see this display:
Login:
4.
Type the following command:
public
8
Dgw v2.0 Application
Accessing the CLI
Software Configuration Guide
Do not type a password, just press <Enter>. After you successfully connect to the Mediatrix unit
through the console port, you can start using the CLI to configure the Mediatrix unit.
If the connection is unsuccessful, check that the cable is properly connected or that you have the
proper cable. Check also the connection parameters of your terminal software.
Accessing the CLI via a Telnet Session
Connecting via Telnet requires a computer with a Telnet remote client running on a PC that acts as a Telnet
host. The Telnet host accesses the Mediatrix unit via its LAN or WAN network interface.
 To access the CLI from a remote host using Telnet:
1.
Set up the Mediatrix unit as described in the Hardware Installation Guide.
2.
Power on your Mediatrix unit. Wait 60 seconds before proceeding to the next step.
3.
Open a Telnet session to the Mediatrix unit (Mediatrix 3000 Series only):
a.
You can use the Mediatrix unit WAN IP address if you know it.
b.
You can use the default Mediatrix unit LAN IP address 192.168.0.10 if the LAN interface is
enabled.
This procedure also requires some knowledge of how to configure the network settings of your
desktop PC. When the unit ships from Media5, it has been assigned a default LAN IP address
of 192.168.0.10. In order to log in to the unit across a network connection, you must use a
machine located on the 192.168.0.0 subnet, or provide a static route in the routing table of your
PC to reach the 192.168.0.0 subnet.
If you are using a Telnet port other than 23, (as configured in “Configuring the CLI” on page 7) you
must also specify it.
4.
Open a Telnet session to the Mediatrix unit by using one of the following IP addresses (Mediatrix
4100/4400/C7 Series):
•
obtained dynamically from the DHCP server
•
you have configured statically
•
after performing a partial reset (192.168.0.1)
•
the link-local IPv6 available and printed on the sticker under the Mediatrix unit (see
“Chapter 11 - IPv4 vs. IPv6” on page 49 for more details)
If you are using a Telnet port other than 23, (as configured in “Configuring the CLI” on page 7) you
must also specify it.
5.
When prompted for a login, type the following:
public
Do not type a password, just press <Enter>. After you successfully connect to the Mediatrix unit by
using Telnet, you can start using the CLI to configure the unit.
Opening a Telnet Session with the Unit Manager Network
You can use the Media5 Unit Manager Network (UMN) product to launch a Telnet client session to configure
the parameters of the Mediatrix unit. You can define which Telnet client to use in the UMN.
The Telnet session is opened from the PC where the client application is installed. It thus establishes a direct
connection to the unit. This could cause some problems if the client PC cannot directly access the unit because
of firewall restrictions, etc.
 To open a Telnet session via UMN:
1.
In the UMN, autodetect the Mediatrix unit at one of the IP addresses listed in “Accessing the CLI via
a Telnet Session” on page 9.
Refer to the Unit Manager Network Administration Manual for more details on how to perform this
task.
2.
Dgw v2.0 Application
Right-click the unit for which to open a Telnet session.
9
Chapter 2 - Command Line Interface (CLI)
3.
Accessing the CLI
Select the Open Telnet Session option in the context sensitive menu that opens.
The following window opens:
Figure 2: Telnet Session Login
This window may differ if you are not using the default Windows Telnet client.
Accessing the CLI via a SSH Session
Connecting via a Secure Socket Shell (SSH) session requires a computer with a SSH or OpenSSH compatible
remote shell client running on a PC that acts as a SSH host. All communication between a client and server
is encrypted before being sent over the network, thus packet sniffers are unable to extract user names,
passwords, and other potentially sensitive data.
 To access the CLI from a remote host using SSH:
1.
Set up the Mediatrix unit as described in the Hardware Installation Guide.
2.
Power on your Mediatrix unit. Wait 60 seconds before proceeding to the next step.
3.
Open a SSH session to the Mediatrix unit (Mediatrix 3000 Series only):
a.
You can use the Mediatrix unit WAN IP address if you know it.
b.
You can use the default Mediatrix unit LAN IP address 192.168.0.10 if the LAN interface is
enabled.
This procedure also requires some knowledge of how to configure the network settings of your
desktop PC. When the Mediatrix unit ships from Media5, it has been assigned a default LAN
IP address of 192.168.0.10. In order to log in to the unit across a network connection, you must
use a machine located on the 192.168.0.0 subnet, or provide a static route in the routing table
of your PC to reach the 192.168.0.0 subnet.
If you are using a SSH port other than 22, (as configured in “Configuring the CLI” on page 7) you
must also specify it.
4.
Open a SSH session to the Mediatrix unit by using one of the following IP addresses (Mediatrix
4100/4400/C7 Series):
•
obtained dynamically from the DHCP server
•
you have configured statically
•
after performing a partial reset (192.168.0.1)
If you are using a SSH port other than 22, (as configured in “Configuring the CLI” on page 7) you
must also specify it.
5.
When prompted for a login, type the following:
public
Do not type a password, just press <Enter>. If you are accessing the unit through the CLI for the
first time or after a factory reset, you may be presented with a warning message regarding the unit’s
identification. You can accept the message and continue.
After you successfully connect to the Mediatrix unit by using Telnet, you can start using the CLI to
configure the Mediatrix unit.
10
Dgw v2.0 Application
Working in the CLI
Software Configuration Guide
Working in the CLI
Refer to the CLI/Conf Scripting Language Syntax on the Documentation Portal.
Dgw v2.0 Application
11
Chapter 2 - Command Line Interface (CLI)
12
Working in the CLI
Dgw v2.0 Application
Introduction
C
Software Configuration Guide
H A P T E R
3
Web Interface Configuration
The Mediatrix unit contains an embedded web server to set parameters by using the HTTP or HTTPS protocol.
This chapter describes the following:



Introduction to the Mediatrix unit web pages.
How to access the web interface and description of the various menus available.
How to submit changes.
Introduction
The web interface may be used to:




View the status of the Mediatrix unit.
Set the uplink parameters of the Mediatrix unit.
Perform a firmware update, configuration scripts download, or configuration backup/restore.
Set numerous parameters of the Mediatrix unit.
All of the parameters in the web interface may also be configured via SNMP. See “Chapter 50 - SNMP
Configuration” on page 509 for more details.
 To configure the web-based configuration service:
1.
In the webMIB, locate the serverGroup folder.
2.
Define the HTTP mode(s) to which the Web server should listen in the httpMode variable.
You can also use the following line in the CLI or a configuration script:
web.httpMode="Value"
where Value may be as follows:
Table 3: HTTP Modes
Value
Mode
Description
100
Secure
The Web server only accepts requests using HTTPS. Requests using HTTP
are ignored. This is the default value.
200
Unsecure The Web server only accepts requests using HTTP. Requests using HTTPS
are ignored.
300
Both
The Web server accepts requests using HTTP or HTTPS.
If you are using HTTPS (either in “Secure” mode or “Both” mode), the web server needs a valid
server certificate with “server authentication” extended key usage installed on the Mediatrix unit.
See “Chapter 49 - Certificates Management” on page 501 for more details.
Accessing the web pages via HTTPS adds additional delay since encryption is used. To access the
unit via HTTPS, your browser must support RFC 2246 (TLS 1.0).
Note that the web server does not listen to the configured modes when the management interface
is down or a configuration error occurred (e.g., missing or invalid certificate for HTTPS mode) while
setting up the web server.
3.
Set the TCP port on which the web service listens for HTTP requests in the serverPort variable.
You can also use the following line in the CLI or a configuration script:
Dgw v2.0 Application
13
Chapter 3 - Web Interface Configuration
Introduction
web.serverPort="Value"
4.
Set the port on which the web service listens for HTTPS requests in the secureServerPort
variable.
You can also use the following line in the CLI or a configuration script:
web.secureServerPort="Value"
5.
Define the allowed cipher suites for the network security settings to which the Web server should
listen when using the HTTPS mode in the httpsCipherSuite variable.
Any connection attempts to the web server using a cipher that is not allowed by the cipher suite will
result in a failure to establish the connection.
You can also use the following line in the CLI or a configuration script:
web.httpsCipherSuite="Value"
where Value may be as follows:
Table 4: HTTPS Cipher Suite Values and Parameters
Value Parameter
100
200
14
CS1
CS2
Description
The Web server only accepts requests using cipher suites:
•
TLS_DHE_RSA_WITH_AES_256_CBC_SHA
•
TLS_DHE_DSS_WITH_AES_256_CBC_SHA
•
TLS_RSA_WITH_AES_256_CBC_SHA
•
TLS_DHE_RSA_WITH_AES_128_CBC_SHA
•
TLS_DHE_DSS_WITH_AES_128_CBC_SHA
•
TLS_RSA_WITH_AES_128_CBC_SHA
•
TLS_DHE_RSA_WITH_3DES_EDE_CBC_SHA
•
TLS_DHE_DSS_WITH_3DES_EDE_CBC_SHA
•
TLS_RSA_WITH_3DES_EDE_CBC_SHA
•
TLS_RSA_WITH_RC4_128_SHA
•
TLS_RSA_WITH_RC4_128_MD5
•
TLS_DHE_RSA_WITH_DES_CBC_SHA
•
TLS_RSA_WITH_DES_CBC_SHA
•
TLS_DHE_RSA_EXPORT_WITH_DES40_CBC_SHA
•
TLS_DHE_DSS_EXPORT_WITH_DES40_CBC_SHA
•
TLS_RSA_EXPORT_WITH_DES40_CBC_SHA
•
TLS_RSA_EXPORT_WITH_RC4_40_MD5
This represents a secure configuration using SHA-1. Web server only
accepts requests using cipher suites:
•
TLS_RSA_WITH_AES_128_CBC_SHA
•
TLS_RSA_WITH_AES_256_CBC_SHA
•
TLS_RSA_WITH_3DES_EDE_CBC_SHA
•
TLS_DHE_RSA_WITH_AES_128_CBC_SHA
•
TLS_DHE_RSA_WITH_AES_256_CBC_SHA
•
TLS_DHE_RSA_WITH_3DES_EDE_CBC_SHA
Dgw v2.0 Application
Introduction
Software Configuration Guide
Table 4: HTTPS Cipher Suite Values and Parameters (Continued)
Value Parameter
300
CS3
Description
This represents the most secure configuration using SHA-2. Only the most
secure cipher suites are allowed when using this value.
•
TLS_ECDHE_RSA_WITH_AES_256_GCM_SHA384
•
TLS_ECDHE_RSA_WITH_AES_256_CBC_SHA384
•
TLS_DHE_RSA_WITH_AES_256_GCM_SHA384
•
TLS_DHE_RSA_WITH_AES_256_CBC_SHA256
•
TLS_ECDHE_RSA_WITH_AES_128_CBC_SHA256
•
TLS_ECDH_RSA_WITH_AES_256_GCM_SHA384
•
TLS_ECDH_RSA_WITH_AES_256_CBC_SHA384
•
TLS_RSA_WITH_AES_256_GCM_SHA384
•
TLS_RSA_WITH_AES_256_CBC_SHA256
•
TLS_ECDHE_RSA_WITH_AES_128_GCM_SHA256
•
TLS_DHE_RSA_WITH_AES_128_GCM_SHA256
•
TLS_DHE_RSA_WITH_AES_128_CBC_SHA256
•
TLS_ECDH_RSA_WITH_AES_128_GCM_SHA256
•
TLS_ECDH_RSA_WITH_AES_128_CBC_SHA256
•
TLS_RSA_WITH_AES_128_GCM_SHA256
•
TLS_RSA_WITH_AES_128_CBC_SHA256
Table 5: Cipher Suites Configuration Values
Value
Meaning
100
CS1
200
CS2
300
CS3
Tls Version Settings
You can define the allowed TLS versions for the network security settings when using the HTTPS. Any
connection attempts to the web server using a TLS version that is not allowed will result in a failure to establish
the connection.
You can configure this parameter as follows:



by using a MIB browser
by using the CLI
by creating a configuration script containing the configuration variables
Table 6: Tls Version Configuration Settings
Parameter
Dgw v2.0 Application
Description
SSLv3
Allow SSL version 3 and all TLS versions.
TLSv1
Allow TLS versions 1 and up.
TLSv1_1
Allow TLS versions 1.1 and up.
15
Chapter 3 - Web Interface Configuration
Table 6: Tls Version Configuration Settings
Parameter
Description
TLSv1_2
Allow TLS versions 1.2 and up.
The default value is TLS1v.
 To set the Tls Version configuration parameter:
1.
In the webMIB, locate the ServerGroup folder.
2.
Set the Tls Version configuration in the tlsVersion parameter.
You can also use the following line in the CLI or a configuration script:
Web.TlsVersion ="Value"
where value may be:
Table 7: Tls Version Configuration Values
Value
Meaning
100
SSLv3
200
TLSv1
300
TLSv1_1
400
TLSv1_2
HTTP User-Agent Header Format
This section describes configuration that is available only in the MIB parameters of the Mediatrix unit. You can
configure these parameters as follows:



by using a MIB browser
by using the CLI
by creating a configuration script containing the configuration variables
You can define the text to display in the HTTP User-Agent header. You can use macros to include information
specific to the unit.
You can also define the same information in the SIP User-Agent header. See “SIP User-Agent Header Format”
on page 297 for more details.
 To set the HTTP User-Agent header format:
1.
In the hocMIB, set the HTTP User-Agent header format in the httpUaHeaderFormat variable.
You can also use the following line in the CLI or a configuration script:
hoc.httpUaHeaderFormat="Value"
where Value may contain any text, as well as one or more of the following macros:
Table 8: Macros Supported
Macro
Description
%version%
Application version.
%mac%
MAC address.
%product% Product name.
16
Dgw v2.0 Application
Software Configuration Guide
Table 8: Macros Supported (Continued)
Macro
Description
%profile%
Profile.
%%
Insert the % character.
For instance, the default value is:
%product%/v%version% %profile%
Dgw v2.0 Application
17
Chapter 3 - Web Interface Configuration
Using the Web Interface
Using the Web Interface
Media5 recommends that you use the latest version of the Microsoft
properly access the web interface.
® Internet Explorer web browser to
 To use the web interface configuration:
1.
In your web browser’s address field, type the IP address of the Mediatrix unit LAN interface (if you
have performed a partial reset, this is 192.168.0.10).
Figure 3: Login Window
2.
Enter the proper user name and password.
The user name and password are case sensitive hence they must be entered properly. Default
factory values are:
•
User Name: admin
•
Password: administrator
You can also enter the user name public and no password.
3.
Click Login.
The Information web page displays. It stays accessible for as long as the Internet browser used to
access the Mediatrix unit web interface is opened.
Figure 4: Information Web Page
18
Dgw v2.0 Application
Submitting Changes
Software Configuration Guide
The Installed Hardware section (Mediatrix 3000 Series only) lists the cards installed in the Mediatrix
unit:
Table 9: Hardware Codes Description
Card Number
Mediatrix 3301-010
Quantity Description
1
FXS/FXO
card
Mediatrix 3308, Mediatrix 3716, Mediatrix 3731,
or Mediatrix 3741
2
FXS/FXO
card
Mediatrix 3316
1
E1/T1 card
Mediatrix 3531, Mediatrix 3621, Mediatrix 3631,
Mediatrix 3731, or Mediatrix 3734
2
E1/T1 card
Mediatrix 3532 or Mediatrix 3632
1
BRI card
Mediatrix 3404, Mediatrix 3734, or Mediatrix
3741
2
BRI card
Mediatrix 3408
1
2 FXS/6
FXO card
Mediatrix 3208, Mediatrix 3716, Mediatrix 3732,
or Mediatrix 3742
2
2 FXS/6
FXO card
Mediatrix 3216
Mediatrix 3301-020
Mediatrix 3301-060
Mediatrix 3301-080
4.
Products
Click Sign Out to end your Mediatrix web session.
The Login Window web page displays.
Submitting Changes
When you perform changes in the web interface and click the Submit button, the Mediatrix unit validates the
changes. A message is displayed next to any invalid value. A message is also displayed if a service must be
restarted and a link is displayed at the top of the page. This link brings you to the Services page. In this page,
each service that requires to be restarted has a “*” beside its name. See “Chapter 4 - Services” on page 23
for more details.
If you are not able to restart one or more services, click the Reboot link in the top menu. The Reboot page then
opens. You must click Reboot. This restarts the Mediatrix unit. If the unit is in use when you click Reboot, all
calls are terminated.
Dgw v2.0 Application
19
Chapter 3 - Web Interface Configuration
20
Submitting Changes
Dgw v2.0 Application
System Parameters
Page Left Intentionally Blank
Services Table
C
Software Configuration Guide
H A P T E R
4
Services
This chapter describes how to view and start/stop system and network parameters of the Mediatrix unit.
Services Table
The Mediatrix unit uses many services grouped in two classes: system and user. You can perform service
commands on user services, but not the system services.
Whenever you perform changes in the various sections of the web interfaces, this usually means that you must
restart a service for the changes to take effect. When a service needs to be restarted, it is displayed in bold
and the message Restart needed is displayed in the Comment column.
If you are not able to restart a service because it is a system service, click the Reboot link in the top menu.
The Reboot page then opens. You must click Reboot. This restarts the Mediatrix unit. If the unit is in use when
you click Reboot, all calls are terminated.
 To manage the Mediatrix unit services:
1.
In the web interface, click the System link, then the Services sub-link.
Figure 5: System – Services Web Page
2
Dgw v2.0 Application
23
Chapter 4 - Services
Services Table
The following are the services available. Note that the services available may differ depending on
the Mediatrix unit you are using.
Table 10: Mediatrix unit Services
Service
Description
System Services
Authentication, Authorization
and Accounting (AAA)
Authenticates a user and grants rights to perform specific tasks
on the system.
Certificate Manager (CERT)
Manages certificate files and provides access to these
certificates.
Configuration Manager
(CONF)
Responsible of configuration scripts transfers, as well as
configuration image upload/download for backup/restore of the
unit configuration.
Device Control Manager
(DCM)
Auto-detects and identifies the hardware components of the unit.
Ethernet Manager (ETH)
Configures the system's Ethernet ports parameters.
File Manager (FILE)
Manages the files created with the File transfer protocol.
Firmware Pack Updater
(FPU)
Handles firmware upgrade and downgrade operations.
Host Configuration (HOC)
Configures network parameters that apply to the Mediatrix unit
(not to a specific interface).
Local Quality Of Service
(LQOS)
Configures the packets tagging sent from the Mediatrix unit.
Process Control Manager
(PCM)
Responsible to boot and restart the unit.
Service Controller Manager
(SCM)
Responsible to:
•
Manage services information.
•
Offer proxy functionality for service interoperation.
User Services
Basic Network Interface
(BNI)
Configures the IP address and network mask for the Uplink and
LAN1 networks.
Call Routing (CROUT)
Routes calls between interfaces.
Call Detail Record (CDR)
24
Command Line Interface
(CLI)
Allows you user to configure the unit parameters by Serial port
(Mediatrix 3000 Series only), Telnet or SSH.
CPE WAN Management
Protocol (CWMP)
Manages the support of the TR-069 protocol for auto
provisioning.
DHCP Server (Dhcp)
Allows the user to lease IP addresses and send network
configuration to hosts located on any network.
Endpoint Administration
(EpAdm)
Holds basic administration and status at endpoint and unit level.
Endpoint Services (EpServ)
Manages endpoint behaviour and holds configuration
parameters related to endpoints (such as DTMF maps,
telephony services, etc.).
IP Routing (IpRouting)
Allows the user to configure the unit's routing table.
Dgw v2.0 Application
Services Table
Software Configuration Guide
Table 10: Mediatrix unit Services (Continued)
Service
2.
Description
Integrated Services Digital
Network (ISDN)
Configures the Integrated Services Digital Network (ISDN) Basic
Rate Interfaces (BRI) or Primary Rate Interfaces (PRI)
parameters of the Mediatrix unit.
Local Firewall (LFW)
Allows you to filter incoming packets whose final destination is
the unit.
Link Layer Discovery
Protocol (Lldp):
Used by network devices for advertising their identity,
capabilities, and neighbors on a IEEE 802 local area network,
usually wired Ethernet.
Media IP Transport (MIPT)
Holds basic configuration parameters (such as voice/data
codec) and implements basic functionality related to media
stream.
Music on Hold (MOH)
Allows you to configure the Music on Hold parameters.
Network Address Translation
(Nat)
Allows the user to change the source or destination address/port
of a packet.
Network Firewall (Nfw)
Allows the user to filter forwarded packets.
Notifications and Logging
Manager (NLM)
Handles syslog messages and notification messages.
Network Traffic Control (Ntc)
Controls the bandwidth limitation applied to physical network
interfaces.
Plain Old Telephony System
Lines service (POTS)
Holds basic configuration parameters (such as DTMF dialing
delays) and implements basic functionality related to POTS lines
(such as enabling/disabling individual lines).
SIP Endpoint (SipEp)
Manages the behaviour of the system regarding SIP.
SNMP (SNMP)
Accesses internal variables through an SNMP client. It also
handles user authentication.
Telephony Interface (TELIF)
Configures the basic specification of each telephony interface.
Web (WEB)
Allows accessing the unit through web pages, using HTTP.
In the User Service section, select the service startup type of a service in the Startup Type column.
Table 11: Startup Types
Type
Auto
Description
The service is automatically started when the system starts.
Manual The administrator must manually start the service.
You can put only user services in manual startup type. Proceed with caution when setting services
to manual because this could prevent you from successfully contacting the unit.
3.
Select if you want to perform service commands on one or more services in the Action column.
Table 12: Actions
Action
Description
Starts the service.
Stops the service.
Dgw v2.0 Application
25
Chapter 4 - Services
Services Table
Table 12: Actions
Action
Description
Restarts the service.
When a service needs to be restarted to apply new configuration you have set elsewhere in the web
interface, it is displayed in bold and the message Restart needed is displayed in the Comment
column.
If you stop, start or restart a service, any dependent services are also affected. The tabs of the
services that have been stopped or have never been started because their startup type is manual
are greyed out. Upon clicking these tabs, a list of services that must be restarted is displayed.
4.
Click the Restart Required Services button at the bottom of the page.
Graceful Restart of Services
You can set a delay to allow for telephony calls to be all completed before restarting services that need a
restart.
During that delay, it is impossible to make new calls but calls in progress are not terminated. When all calls
are completed, then the restart is authorized and the services that require a restart are restarted.
You can also set a unit restart grace period when performing a Firmware Upgrade as described in “Firmware
Download” on page 499.
 To configure the graceful restart of services:
1.
In the Restart Required Services section, set the Graceful Delay field with the delay (in minutes)
allowed for telephony calls to be all completed.
At the expiration of this delay, the services are forced to restart.
Figure 6: Services – Restart Required Services Section
1
2.
Click Restart Required Services to restart only the services that needed a restart for their
configuration to be applied.
If you click Cancel, this cancels the restart during the grace delay period.
Restarting a Service via MIB
If you are using a MIB browser to access the Mediatrix unit configuration via SNMP, you can determine
whether or not a service needs to be restarted by locating the configurationGroup folder of the related service
and checking if the service needs to be restarted in the needRestartInfo variable.
Figure 7: Need Restart Info
If a specific service needs to be restarted, locate the scmMIB, then set the serviceCommandsRestart variable
for this service to restart.
26
Dgw v2.0 Application
Services Table
Software Configuration Guide
Figure 8: Restart Service
You can also start a service by setting the serviceCommandsStart variable for this service to Start.
You can also stop a service by setting the serviceCommandsStop variable for this service to Stop.
If you are not able to restart a service because it is a system service, you must restart the Mediatrix unit.
Dgw v2.0 Application
27
Chapter 4 - Services
28
Services Table
Dgw v2.0 Application
Software Configuration Guide
C
H A P T E R
5
Hardware Parameters
Refer to the Hardware Configuration User Guide located on the http://www.media5corp.com/
documentation.
Dgw v2.0 Application
29
Chapter 5 - Hardware Parameters
30
Dgw v2.0 Application
Unit Configuration
C
Software Configuration Guide
H A P T E R
6
Endpoints State Configuration
This chapter describes how to set the administrative state of the Mediatrix unit’s endpoints.
Unit Configuration
The unit configuration section allows you to define the administrative state of all the Mediatrix unit’s endpoints.
 To set the unit’s endpoints parameters:
1.
In the web interface, click the System link, then the Endpoints sub-link.
Figure 9: System Configuration – Endpoints Web Page
2
2.
In the Unit States section, select a temporary state for all of the unit’s endpoints in the Action
column.
This command locks/unlocks all endpoints of the Mediatrix unit. This state is kept until you modify
it or the unit restarts. It offers the following settings:
Table 13: Action Settings
Setting
3.
Description
Force Lock
Cancels all the endpoints registration to the SIP server. All active calls in
progress are terminated immediately. No new calls may be initiated.
Lock
Cancels all the endpoints registration to the SIP server. Active calls in
progress remain established until normal call termination. No new calls
may be initiated.
Unlock
Registers the endpoints to the SIP server.
If you do not need to set other parameters, click Submit.
Endpoints Configuration
The endpoints configuration allows you to define the administrative state of the Mediatrix unit’s endpoints.
Dgw v2.0 Application
31
Chapter 6 - Endpoints State Configuration
Administration
 To set the endpoints parameters:
1.
In the Endpoint States section of the Endpoints page, select the permanent administrative state
each endpoint will have when the Mediatrix unit restarts in the Initial Administrative column.
Figure 10: Endpoint States Section
1
2
Table 14: Permanent Administrative State Settings
Setting
2.
Description
Unlocked
Registers the endpoint to the SIP server.
Locked
The endpoint is unavailable for normal operation. It cannot be used to
make and/or receive calls.
Select a temporary state for each endpoint in the corresponding Action column.
This command locks/unlocks an endpoint of the Mediatrix unit. This state is kept until you modify it
or the unit restarts. It offers the following settings:
Table 15: Action Settings
Setting
3.
Description
Force Lock
Cancels the endpoint registration to the SIP server. All active calls in
progress are terminated immediately. No new calls may be initiated.
Lock
Cancels the endpoint registration to the SIP server. Active calls in
progress remain established until normal call termination. No new calls
may be initiated.
Unlock
Registers the endpoint to the SIP server.
If you do not need to set other parameters, click Submit.
Administration
The Administration section allows you to define endpoint operational state.
 To set administration parameters:
1.
In the Administration section of the Endpoints page, set the Disable Unit (All Endpoints) When No
Gateways Are In State Ready drop-down menu with the proper behaviour.
Figure 11: Administration Section
1
2
Table 16: Unit Operational State Parameters
Parameter
Disable
32
Description
Signaling gateways have no impact on the unit operational
state
Dgw v2.0 Application
Administration
Software Configuration Guide
Table 16: Unit Operational State Parameters (Continued)
Parameter
Description
Enable
2.
When all signaling gateways are not ready, the unit
operational state is set to disabled.
Set the Shutdown Endpoint When Operational State is Disable And Its Usage State Is 'idleunusable' drop-down menu with the proper behaviour.
Table 17: Endpoint Shutdown Parameters
Parameter
Description
Enable
When the usage state becomes “Idle-unusable” and the
operational state becomes “Disable”, the endpoint is
physically shutdown.
Disable
When an endpoint's usage state becomes “Idle-unusable”
whatever the value of its operational state, the endpoint
remains physically up but the calls are denied.
The default value is:
3.
•
Enable for the Mediatrix LP/4100/C7 series
•
Disable for the Mediatrix 3000 and Mediatrix 4400 series
Click Submit if you do not need to set other parameters.
Unit Shutting Down Behaviour
This section describes configuration that is available only in the MIB parameters of the Mediatrix unit. You can
configure these parameters as follows:



by using a MIB browser
by using the CLI
by creating a configuration script containing the configuration variables
You can configure the behaviour of the call permissions when the UnitAdminState is ShuttingDown.
The following parameters are available:
Table 18: Unit Shutting Down Behaviour Parameters
Parameter
Description
BlockNewCalls No new requests are accepted once all activity are terminated. Endpoints cannot make and
receive calls.
AllowNewCalls New requests are accepted until all activities are simultaneously terminated. Endpoints can
make and receive calls.
 To set the unit shutting down behaviour:
1.
In the epAdmMIB, locate the UnitConfigGroup folder.
2.
Set the behaviorWhileInUnitShuttingDownState variable with the proper behaviour.
You can also use the following line in the CLI or a configuration script:
epAdm.behaviorWhileInUnitShuttingDownState="Value"
Dgw v2.0 Application
33
Chapter 6 - Endpoints State Configuration
Administration
where Value may be one of the following:
Table 19: Unit Shutting Down Behaviour Values
Value
34
Meaning
100
BlockNewCalls
200
AllowNewCalls
Dgw v2.0 Application
Syslog Daemon Configuration
C
Software Configuration Guide
H A P T E R
7
Syslog Configuration
This chapter describes how the Mediatrix unit handles syslog messages and notification messages.
For a list and description of all syslog messages and notification messages that the Mediatrix unit may send,
refer to the Notification Reference Guide.
Syslog Daemon Configuration
The Syslog daemon is a general purpose utility for monitoring applications and network devices with the TCP/
IP protocol. With this software, you can monitor useful messages coming from the Mediatrix unit unit. If no
Syslog daemon address is provided by a DHCP server or specified by the administrator, no messages are
sent.
For instance, if you want to download a new firmware into the Mediatrix unit, you can monitor each step of the
firmware download phase. Furthermore, if the unit encounters an abnormal behaviour, you may see accurate
messages that will help you troubleshoot the problem.
The Mediatrix unit supports RFC 3164 as a “device” only (see definition of device in section 3 of the RFC).
Dgw v2.0 Application
35
Chapter 7 - Syslog Configuration
Syslog Daemon Configuration
 To configure the Mediatrix unit syslog client:
1.
In the web interface, click the System link, then the Syslog sub-link.
Figure 12: System – Syslog Web Page
2
3
4
5
2.
Set the static IP address or domain name and port number of the device to use to archive log entries
in the Remote Host field.
Use the special port value zero to indicate the protocol default. For instance, the TFTP default port
is 69 and the HTTP/HTTPS default port is 80.
3.
In the Service Severity section, select the minimal severity to issue a notification message for the
various services in the corresponding drop-down menus.
Any syslog message with a severity value greater than the selected value is ignored. Available
values are:
Table 20: Severity Values
Severity
36
Description
Notification Messages Issued
Disable
N/A
No notification is issued.
Debug
Message describing in detail the unit's
operations.
All notification messages are issued.
Info
Message indicating a significant event for
the unit's normal operations.
Notification messages with severity
“Informational” and higher are issued.
Warning
Message indicating an abnormal event or
situation that could be potentially risky.
The unit may not be fully operational.
Notification messages with severity
“Warning” and higher are issued.
Dgw v2.0 Application
Syslog Daemon Configuration
Software Configuration Guide
Table 20: Severity Values (Continued)
Severity
Description
Notification Messages Issued
Error
Message indicating an abnormal event or Notification messages with severity
situation, the system's operation is
“Error” and higher are issued.
affected. The unit may not be operational.
Critical
Message indicating a critical event or
situation that requires immediate
attention. The unit is not operational.
Notification messages with severity
“Critical” are issued.
A higher level mask includes lower level masks, e.g., Warning includes Error and Critical. The
default value is Warning.
4.
In the Technical Assistance Centre section, enable diagnostic traces by setting the Diagnostic
Traces drop-down menu to Enable.
At the request of Media5’s Technical Support personnel, enabling these traces will allow Media5 to
further assist you in resolving some issues. However, be advised that enabling this feature issues
a lot of messages to the syslog host. These messages may be filtered by using the Diagnostic
Traces Filter field.
Note: Enabling all the traces could affect the performance of the Mediatrix unit.
5.
If applicable, define the filter applied to diagnostic traces by clicking the Edit button in the Filter field.
The following opens:
Figure 13: Diagnostic Traces Window
You can use the filter to narrow down the number of traces sent at the request of Media5’s Technical
Support personnel.
6.
Dgw v2.0 Application
Click Submit if you do not need to set other parameters.
37
Chapter 7 - Syslog Configuration
Syslog Daemon Configuration
Configuring PCM Capture
Refer to the Dgw PCM Traces technical bulletin on our Documentation Portal.
Configuring the Syslog Daemon Application
You must configure the Syslog daemon server to capture those messages. Refer to your Syslog daemon’s
documentation to learn how to properly configure it to capture messages.
38
Dgw v2.0 Application
Notification Events
C
Software Configuration Guide
H A P T E R
8
Events Configuration
This chapter describes how to associate a NOTIFICATION message and how to send it (via syslog or via a
SIP NOTIFY packet).
For a list and description of all syslog messages and notification messages that the Mediatrix unit may send,
refer to the Notification Reference Guide.
Notification Events
You can configure an event router in order to apply a set of rules to select the proper transport protocol
scheme. A rule entry is made up of three different values: type, criteria and action.
Note that more than one notification may be sent for a single event based on the event router table rules.
 To configure notification events:
1.
Ensure that the severity level for all services are set according to the severity level of the notification
messages that are required by the system administrator. See “Chapter 7 - Syslog Configuration” on
page 35 for more details.
2.
In the web interface, click the System link, then the Events sub-link.
Figure 14: System – Events Web Page
4
5
6
7
8
3
3.
If you want to add a rule entry before an existing entry, locate the proper row in the table and click
the
button of this row.
4.
Set the Activation drop-down menu with the current activation state for the corresponding system
event.
Table 21: Activation Parameters
Parameter
5.
Description
Enable
This action is enabled for this system event.
Disable
This action is disabled for this system event.
Optional: Set the corresponding Criteria field with the expression an event must match in order to
apply the specified action. The expression is based on the event type.
This step is optional because a proper value may be automatically entered by the Mediatrix unit
upon setting the Service (Step 5) and Notification (Step 6) drop-down menus.
An event of type notification uses the notification ID as expression criteria. The notification ID is the
combination of the service number key and the message number key separated by a dot. The
information regarding the service and message number key is available in the Notification
Reference Guide document.
Dgw v2.0 Application
39
Chapter 8 - Events Configuration
Notification Events
Several basic criteria can also be specified on the same line, separated by commas. Criteria can
specify inclusion or exclusion. A group of exclusion criteria can follow the group of inclusion criteria.
The group of exclusion criteria must begin with a hyphen (-).
Matching an inclusion criteria causes the action to be executed unless an exclusion criteria is also
matched. Exclusion criteria have precedence over inclusion criteria.
Spaces are allowed before or after a basic criterion; however, spaces are not accepted within a
basic criterion, i.e. before or after the dot.
Examples:
Service ISDN (number key = 1850)
Message %1$s: Physical link state changed to up (number key = 5)
The corresponding Criteria is: 1850.5
You can also use the special expression All, which means all available services and messages.
Criteria 1850.All,1600.200,1600.W,-1850.500,1600.300
1850.All,1600.200,1600.W are inclusion criteria and -1850.500,1600.300 are exclusion criteria. All
notifications from service 1850, except notification 500, will match the expression. All notifications
from service 1600 with Warning level, except notification 300, will match the expression. Notification
200 from service 1600 will match the expression, no matter the severity level.
6.
In the corresponding Service drop-down menu, select the service for which you want to send
events.
You can use the Suggestion column’s drop-down menu to select between suggested values, if any.
7.
In the Notification drop-down menu, select the notification message that you want to send.
You can use the Suggestion column’s drop-down menu to select between suggested values, if any.
8.
In the Action drop-down menu, select the action to apply to the system event if the criteria matches.
The action represents a transport targeted for the event. The format of the event under which the
message is carried is dependent on the protocol in use.
The possible actions are:
Table 22: Action Parameters
Parameter
Description
Send Via
Syslog
The event notification is sent using syslog as transport. See “Chapter 7 - Syslog
Configuration” on page 35 for more details.
Send Via
SIP
The event notification is sent using SIP Notify as transport.
Log
Locally
Log is stored in the volatile memory (RAM) and displayed on the Web page/
SNMP/TR-069
Log to File
Log is stored in a file in the persistent storage (flash) and available through file
transfers.
Note: The Log to File action is only available on units with 1024 KB or more of persistent storage i.e. on
Mediatrix Sentinel, Mediatrix 44XX, Mediatrix LP 16/24, Mediatrix 4108/4116/4124, and Mediatrix 3000.
9.
Click the Apply button.
The configuration status of the row displays on the right part of the row. It indicates whether the
configuration of the row is valid.
Table 23: Configuration Status Values
Value
Valid
40
Description
The current content of the fields Type, Criteria and Action is valid.
Dgw v2.0 Application
Notification Events
Software Configuration Guide
Table 23: Configuration Status Values (Continued)
Value
Description
Invalid
The current content of the fields Type, Criteria and Action is not valid.
Not
Supported
The current content of the fields Type, Criteria and Action is valid but not
supported.
Deleting a Rule
You can delete a rule row from the table in the web interface.
 To delete a rule entry:
1.
Click the
button of the row you want to delete.
2.
Click the Apply button.
Monitoring Parameters
This section describes configuration that is available only in the MIB parameters of the Mediatrix unit. You can
configure these parameters as follows:



by using a MIB browser
by using the CLI
by creating a configuration script containing the configuration variables
You can set two monitoring parameters for the Notification Events table.
 To set monitoring parameters:
1.
In the sipEpMIB, locate the MonitoringGroup folder.
2.
Set the sipNotificationsGateway variable with the SIP gateway used to send SIP NOTIFY
containing the notification events.
You can also use the following line in the CLI or a configuration script:
sipEp.sipNotificationsGateway="Value"
Value is the name of the SIP gateway from which the NOTIFICATION is sent.
3.
Set the maxNotificationsPerNotify variable with the maximal number of notification events the
device may have to send in one SIP NOTIFY request.
Notifications are sent in XML elements through the SIP NOTIFY's body request.
You can also use the following line in the CLI or a configuration script:
sipEp.maxNotificationsPerNotify="Value"
Value may be between 1 and 25.
Dgw v2.0 Application
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Chapter 8 - Events Configuration
42
Notification Events
Dgw v2.0 Application
Local Log Status and Entries
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H A P T E R
9
Local Log
This chapter describes local log status and entries generated by the Notifications and Logging Manager (NLM)
for your Mediatrix unit.
Local Log Status and Entries
You can display, clear and refresh local log status and entries.
 To manage local log status and entries:
1.
In the web interface, click the System link, then the Local Log sub-link.
Figure 15: System – Local Log Web Page
2
3
The following is the Local Log Status information displayed.
Table 24: Local Log Status Parameters
Parameter
Description
Maximum Number of Entries
Maximum number of entries that the local log can contain. When
adding a new entry while the local log is full, the oldest entry is
erased to make room for the new one.
Number of Error Entries
Current number of error entries in the local log.
Number of Critical Entries
Current number of critical entries in the local log.
The following is the Local Log Entries information displayed.
Table 25: Local Log Entries Parameters
Parameter
Dgw v2.0 Application
Description
Local Time
Local date and time at which the log entry was inserted. Format is
YYYY-MM-DD HH:MM:SS.
Severity
Severity of the log entry.
Service Name
Textual identifier of the service that issued the log entry.
Service Key
Numerical identifier of the service that issued the log entry.
Message Key
Numerical identifier of the notification message.
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Chapter 9 - Local Log
Local Log Status and Entries
Table 25: Local Log Entries Parameters
Parameter
Message Content
44
Description
The readable content of the log message.
2.
Click Clear Local Log to clear all log entries.
3.
Click Refresh Local Log to refresh the log entries display.
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10
VM
This chapter describes how to use the Virtual Machine (VM) service. The VM service allows the
administrator to manage virtual machines.
Note: This web page is available only on the following model:
• Sentinel
For more details, refer to the VM Service User Guide.
Dgw v2.0 Application
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Chapter 10 - VM
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Network Parameters
Page Left Intentionally Blank
Introduction
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H A P T E R
11
IPv4 vs. IPv6
This chapter describes the differences between IPv4 and IPv6 addressing.
Introduction
IPv6 (Internet Protocol version 6) is the successor to the most common Internet Protocol today (IPv4). This is
largely driven by the fact that IPv4’s 32-bit address is quickly being consumed by the ever-expanding sites and
products on the internet. IPv6’s 128-bit address space should not have this problem for the foreseeable future.
IPv6 addresses, in addition to being longer, are distinguished from IPv4 addresses by the use of colons ":",
e.g., 2001:470:8929:4000:201:80ff:fe3c:642f. An IPv4 address is noted by 4 sets of decimal numbers
separated by periods ".", e.g., 192.168.10.1.
Please note that IPv6 addresses should be written between [ ] to allow port numbers to be set. For instance:
[fd0f:8b72:5::1]:5060.
IPv4 vs. IPv6 Availability
The Mediatrix unit fully supports IPv4 IP addresses, as well as IPv6 IP addresses in some of its features. The
following table lists all the network related features of the Mediatrix unit with their availability in IPv4 and IPv6.
Table 26: IPv4 vs. IPv6 Availability
Feature
IPv4
IPv6
Backup/Restore transfer
Command Line Interface (CLI)
Configuration file transfer
Embedded DHCP server
Firmware Transfer
IP Routing
Link Layer Discovery Protocol (LLDP) QoS settings
Local Firewall (LFW)
Network Address Translation (NAT)
Network Configuration (IP addresses, DNS and SNTP servers)
Network Firewall (NFW)
Online Certificate Status Protocol (OCSP)
Remote Authentication Dial In User Service (Radius )
SIP signaling and media transport
Simple Network Management Protocol (SNMP)
TR-069
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Chapter 11 - IPv4 vs. IPv6
IPv6 Scope Identifier
Table 26: IPv4 vs. IPv6 Availability (Continued)
Feature
IPv4
IPv6
WEB Configuration
If you configure the Mediatrix unit with IPv6 addresses, then decide to go downgrade to a firmware version
that does not support IPv6, all IPv6 networks are deleted.
Please note that IPv6 addresses should be written between [ ]. For instance: [fd0f:8b72:5::1].
IPv6 Scope Identifier
When using an IPv6 address starting with "FE80::" (IPv6 link-local addresses), there must be additional
information: the IPv6 scope identifier (this represents the network link that will be used to contact the IPv6 linklocal address). The format is "[IPv6 link-local%ScopeIdentifier]".
When Contacting the unit using its IPv6 link-local Address
On Windows, the scope identifier is represented by an interface number. The interface number can be
determined through the command line of Windows.


Go to Start -> Run and type cmd to enter the command prompt.
At the command prompt, type ipconfig and find the IPv6 address. Appended to the end of this
will be a "%x" where x is the interface number.
To contact the IPv6 link-local IPv6 address "fe80::201:80ff:fe3c:642f", you would use:
[fe80::201:80ff:fe3c:642f%4]
On Linux, the scope identifier may be the link name or the interface number. The interface number can be
determined through the Linux command line.
To contact the IPv6 link-local IPv6 address "fe80::201:80ff:fe3c:642f", you would use:
[fe80::201:80ff:fe3c:642f%2] or [fe80::201:80ff:fe3c:642f%eth0]
When Configuring the Mediatrix unit to use an IPv6 link-local Address
In that case, the scope identifier represents the "link" in Network/Interfaces.
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IPv6 Scope Identifier
Software Configuration Guide
For instance, if you want your unit to contact a server with the address IPv6 link-local
"fe80::201:80ff:fe3c:642f", you must check on which network link the server is available. Some units have
"wan" or "lan". Let’s say it is on the "wan" link. The IP address whoud then become
"[fe80::201:80ff:fe3c:642f%wan]".
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Chapter 11 - IPv4 vs. IPv6
IPv6 Scope Identifier
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Dgw v2.0 Application
General Configuration
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Host Parameters
This chapter describes how to set the host information of the Mediatrix unit:






General Configuration (automatic configuration interface)
Host name and domain name.
Default gateway parameters.
DNS parameters.
SNTP client parameters.
Time parameters.
General Configuration
The General Configuration section allows you to configure the networks that will provide the automatic
configuration (host name, default gateway, DNS servers and SNTP servers) used by the Mediatrix unit.
Automatic configuration may be provided via IPv4 (DHCPv4) and/or via IPv6 (stateless auto-configuration and
DHCPv6).
 To set the general configuration:
1.
In the web interface, click the Network link, then the Host sub-link.
Figure 16: Network – Host Web Page
2
3
2.
Set the Automatic IPv4 config source network drop-down menu with the IPv4 network interface that
provides the automatic configuration.
3.
Set the Automatic IPv6 config source network drop-down menu with the IPv6 network interface that
provides the automatic configuration.
4.
Click Submit if you do not need to set other parameters.
The current automatic configuration interface is displayed in the Status page.
Host Configuration
The Host Configuration section allows you to configure the host name and domain name of the Mediatrix unit.
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Chapter 12 - Host Parameters
Host Configuration
 To set the host configuration:
1.
In the Host Configuration section of the Host page, select the configuration source of the domain
name information in the Domain Name Configuration Source drop-down menu.
Figure 17: Host Name Configuration Section
2
3
4
Table 27: Host Name Configuration Sources
Source
Description
Automatic The domain name is automatically obtained from the network. The value obtained
IPv4
depends on the connection type of the automatic network interface (see “General
Configuration” on page 53) if any. Using the automatic configuration assumes that
you have properly set your network server with the relevant information.
Note: Some Uplink connection types (for example Static and PPPoE) cannot obtain
domain name information from the network, and therefore lead to no domain name
being applied to the system.
Automatic The domain name is automatically obtained from the IPv6 network defined in the
IPv6
Automatic IPv6 config source network drop-down menu.
Static
You manually enter the domain name and it remains the same every time the
Mediatrix unit restarts. Use the static configuration if you are not using a network
server or if you want to bypass it.
When switching from the Static to Automatic IPv4 or Automatic IPv6 configuration source, the last
value correctly obtained from the network (if any) is applied to the system.
Static Configuration Source Only
2.
Set the system’s domain name in the Domain Name field.
A domain name is a name of a device on the Internet that distinguishes it from the other systems
on the network. For instance: example.com.
3.
Set the system's host name in the Host Name field.
The host name is the unique name by which the device is known on a network. It may contain any
of the following characters:
•
A to Z and a to z letters
•
0 to 9 digits
•
-._~
•
!$&'()*+=
Certain restrictions apply to this name:
•
4.
The host name must be shorter than 64 characters.
•
The host name must not start with a period.
•
The host name must not contain double quotes, semicolons, curly braces, spaces, and
commas.
•
The host name must not contain the following characters: :/?#[@
Click Submit if you do not need to set other parameters.
The current domain name is displayed in the Status page.
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Default Gateway Configuration
Software Configuration Guide
Default Gateway Configuration
The default gateway (also known as default router) is the gateway to which the Mediatrix unit sends packets
when all other internally known routes have failed.
 To set the default gateway configuration:
IPv4 Configuration
1.
In the Default Gateway Configuration – IPv4 section of the Host page, select the IPv4 configuration
source of the default gateway information in the Configuration Source drop-down menu.
Figure 18: Default Gateway Configuration Section
1
2
3
4
Table 28: Default Gateway Configuration Sources
Source
Description
Automatic The default gateway is automatically obtained from the network. The value obtained
IPv4
depends on the connection type of the automatic network interface (see “General
Configuration” on page 53) if any. Using the automatic configuration assumes that
you have properly set your network server with the relevant information.
Note: Some Uplink connection types (for example Static) cannot obtain default
gateway information from the network, and therefore lead to no default gateway being
applied to the system.
Static
You manually enter the IP address of the default gateway and it remains the same
every time the Mediatrix unit restarts. Use the static configuration if you are not using
a network server or if you want to bypass it.
When switching from the Static to Automatic configuration source, the last value correctly obtained
from the network (if any) is applied to the system.
IPv4 Static Configuration Source Only
2.
If the default gateway configuration source is Static, enter the static default gateway address in the
IP address field.
This can be an IP address or domain name. The default value is 192.168.10.10.
IPv6 Configuration
3.
In the Default Gateway Configuration – IPv6 section of the Host page, select the IPv6 configuration
source of the default gateway information in the Configuration Source drop-down menu.
Table 29: IPv6 Default Gateway Configuration Sources
Source
Description
Automatic The default gateway name is automatically obtained from the IPv6 network defined
IPv6
in the Automatic IPv6 config source network drop-down menu.
Static
Dgw v2.0 Application
You manually enter the IPv6 address of the default gateway and it remains the same
every time the Mediatrix unit restarts. Use the static configuration if you are not using
a network server or if you want to bypass it.
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Chapter 12 - Host Parameters
DNS Configuration
When switching from the Static to Automatic IPv6 configuration source, the last value correctly
obtained from the network (if any) is applied to the system.
4.
If the default gateway configuration source is Static, enter the static default gateway IPv6 address
in the IP address field.
This can be an IP address or domain name.
5.
Click Submit if you do not need to set other parameters.
The current default gateway address is displayed in the Status page.
DNS Configuration
You can use up to four Domain Name Servers (DNS) to which the Mediatrix unit can connect. The DNS
servers list is the ordered list of DNS servers that the Mediatrix unit uses to resolve network names. DNS
query results are cached on the system to optimize name resolution time.
 To set the DNS configuration:
1.
In the DNS Configuration section of the Host page, select the configuration source of the DNS
information in the Configuration Source drop-down menu.
Figure 19: DNS Configuration Section
1
2
Table 30: DNS Configuration Sources
Source
Description
Automatic The DNS servers are automatically obtained from the network. The value obtained
IPv4
depends on the connection type of the automatic network interface (see “General
Configuration” on page 53) if any. Using the automatic configuration assumes that
you have properly set your network server with the relevant information.
Note: Some Uplink connection types (for example Static) cannot obtain DNS
information from the network, and therefore lead to no DNS servers being applied to
the system.
Automatic The DNS servers are automatically obtained from the IPv6 network defined in the
IPv6
Automatic IPv6 config source network drop-down menu.
Static
You manually enter up to four DNS servers IP addresses and they remain the same
every time the Mediatrix unit restarts. Use the static configuration if you are not using
a network server or if you want to bypass it.
When switching from the Static to Automatic IPv4 or Automatic IPv6 configuration source, the last
values correctly obtained from the network (if any) are applied to the system.
Static Configuration Source Only
2.
56
If the DNS configuration source is Static, enter up to four static DNS addresses in the following
fields:
•
Primary DNS
•
Secondary DNS
•
Third DNS
•
Fourth DNS
Dgw v2.0 Application
SNTP Configuration
Software Configuration Guide
3.
Click Submit if you do not need to set other parameters.
The current list of DNS servers is displayed in the Status page.
SNTP Configuration
The Simple Network Time Protocol (SNTP) enables the notion of time (date, month, time) into the Mediatrix
unit. SNTP is used to synchronize a SNTP client with a SNTP or NTP server by using UDP as transport. It
updates the internal clock of the unit to maintain the system time accurate. It is required when dealing with
features such as the caller ID.
The Mediatrix unit implements a SNTP version 3 client.
Note: The Mediatrix unit hardware does not include a real time clock. The unit uses the SNTP client to get
and set its clock. As certain services need correct time to work properly (such as HTTPS), you should
configure your SNTP client with an available SNTP server in order to update and synchronise the local clock
at boot time.
 To set the SNTP client of the Mediatrix unit:
1.
In the SNTP Configuration section of the Host page, select the configuration source of the SNTP
information in the Configuration Source drop-down menu.
Figure 20: SNTP Configuration Section
1
2
3
Table 31: SNTP Configuration Sources
Source
Description
Automatic The SNTP parameters are automatically obtained from the network. The value
IPv4
obtained depends on the connection type of the automatic network interface (see
“General Configuration” on page 53) if any. Using the automatic configuration
assumes that you have properly set your network server with the relevant
information.
Note: Some Uplink connection types (for example Static and PPPoE) cannot obtain
SNTP information from the network, and therefore lead to no SNTP parameters being
applied to the system.
Automatic The SNTP parameters are automatically obtained from the IPv6 network defined in
IPv6
the Automatic IPv6 config source network drop-down menu.
Static
You manually enter the values and they remain the same every time the Mediatrix
unit restarts. Use the static configuration if you are not using a network server or if
you want to bypass it.
Automatic The SNTP parameters are automatically obtained from the IPv4 Network from a
with
fallback to the StaticSntpServers table.
Fallback
When switching from the Static to Automatic IPv4 or Automatic IPv6 configuration source, the last
values correctly obtained from the network (if any) are applied to the system.
Static Configuration Source Only
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Chapter 12 - Host Parameters
Time Configuration
2.
3.
If the SNTP configuration source is Static, enter up to four static SNTP server IP addresses or
domain names and port numbers in the following fields:
•
Primary SNTP
•
Secondary SNTP
•
Third SNTP
•
Fourth SNTP
Set the synchronization information:
Table 32: SNTP Synchronization Information
Field
Description
Synchronisation Period
Time interval (in minutes) between system time
synchronization cycles. Each time this interval expires, a
SNTP request is sent to the SNTP server and the result is
used to set the system time. The maximum value is set to
1 440 minutes, which corresponds to 24 hours.
Synchronisation Period on Error Time interval (in minutes) between retries after an
unsuccessful attempt to reach the SNTP server. The
maximum value is set to 1 440 minutes, which corresponds to
24 hours.
4.
Click Submit if you do not need to set other parameters.
The current SNTP host is displayed in the Status page.
Time Configuration
You can define the current system date and time configured in the unit by specifying in which time zone the
unit is located.
If the time seems not valid, verify the SNTP configuration in “SNTP Configuration” on page 57.
 To set the time of the Mediatrix unit:
1.
In the Time Configuration section of the Host page, enter a valid string in the Static Time Zone field.
Figure 21: Time Configuration Section
1
The format of the string is validated upon entry. Invalid entries are refused. The default value is:
EST5DST4,M4.1.0/02:00:00,M10.5.0/02:00:00
A POSIX string is a set of standard operating system interfaces based on the UNIX operating
system. The format of the IEEE 1003.1 POSIX string is defined in the bootp-dhcp-option-88 Internet
draft as:
STDOFFSET[DST[OFFSET],[START[/TIME],END[/TIME]]]
Refer to the following sub-sections for explanations on each part of the string.
2.
Click Submit if you do not need to set other parameters.
The current system time is displayed in the Status page.
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Software Configuration Guide
STD / DST
Three or more characters for the standard (STD) or alternative daylight saving time (DST) time zone. Only STD
is mandatory. If DST is not supplied, the daylight saving time does not apply. Lower and upper case letters are
allowed. All characters are allowed except digits, leading colon (:), comma (,), minus (-), plus (+), and ASCII
NUL.
OFFSET
Difference between the GMT time and the local time. The offset has the format h[h][:m[m][:s[s]]]. If no offset is
supplied for DST, the alternative time is assumed to be one hour ahead of standard time. One or more digits
can be used; the value is always interpreted as a decimal number.
The hour value must be between 0 and 24. The minutes and seconds values, if present, must be between 0
and 59. If preceded by a minus sign (-), the time zone is east of the prime meridian, otherwise it is west, which
can be indicated by the preceding plus sign (+). For example, New York time is GMT 5.
START / END
Indicates when to change to and return from the daylight saving time. The START argument is the date when
the change from the standard to the daylight save time occurs; END is the date for changing back. If START
and END are not specified, the default is the US Daylight saving time start and end dates. The format for start
and end must be one of the following:

n where n is the number of days since the start of the year from 0 to 365. It must contain the
leap year day if the current year is a leap year. With this format, you are responsible to
determine all the leap year details.

Jn where n is the Julian day number of the year from 1 to 365. Leap days are not counted. That
is, in all years – including leap years – February 28 is day 59 and March 1 is day 60. It is
impossible to refer to the occasional February 29 explicitly. The TIME parameter has the same
format as OFFSET but there can be no leading minus (-) or plus (+) sign. If TIME is not
specified, the default is 02:00:00.

Mx[x].y.z where x is the month, y is a week count (in which the z day exists) and z is the day
of the week starting at 0 (Sunday). For instance:
M10.4.0
is the fourth Sunday of October. It does not matter if the Sunday is in the 4th or 5th week.
M10.5.0
is the last Sunday of October (5 indicates the last z day). It does not matter if the Sunday is in the
4th or 5th week.
M10.1.6
is the first week with a Saturday (thus the first Saturday). It does not matter if the Saturday is in the
first or second week.
The TIME parameter has the same format as OFFSET but there can be no leading minus (-) or plus
(+) sign. If TIME is not specified, the default is 02:00:00.
Example
The following is an example of a proper POSIX string:
Standard
time zone
Offset
Month, Week, and Day
to start the Daylight
Saving Time
Month, Week, and Day
to stop the Daylight
Saving Time
EST5DST4,M4.0.0/02:00:00,M10.5.0/02:00:00
Daylight
Saving Time
time zone
Dgw v2.0 Application
Offset
Time to start the
Daylight Saving
Time
Time to stop
the Daylight
Saving Time
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Chapter 12 - Host Parameters
Additional Parameters
The following are some valid POSIX strings:
Table 33: Valid POSIX Strings
Time Zone
POSIX String
Pacific Time (Canada & US)
PST8PDT7,M3.2.0/02:00:00,M11.1.0/02:00:00
Mountain Time (Canada & US)
MST7MDT6,M3.2.0/02:00:00,M11.1.0/02:00:00
Central Time (Canada & US)
CST6CDT5,M3.2.0/02:00:00,M11.1.0/02:00:00
Eastern Time Canada & US)
EST5EDT4,M3.2.0/02:00:00,M11.1.0/02:00:00
Atlantic Time (Canada)
AST4ADT3,M3.2.0/02:00:00,M11.1.0/02:00:00
GMT Standard Time
GMT0DMT-1,M3.5.0/01:00:00,M10.5.0/02:00:00
W. Europe Standard Time
WEST-1DWEST-2,M3.5.0/02:00:00,M10.5.0/03:00:00
China Standard Time
CST-8
Tokyo Standard Time
TST-9
Central Australia Standard Time
CAUST-9:30DCAUST-10:30,M10.5.0/02:00:00,M3.5.0/02:00:00
Australia Eastern Standard Time
AUSEST-10AUSDST-11,M10.5.0/02:00:00,M3.5.0/02:00:00
UTC (Coordinated Universal Time)
UTC0
Additional Parameters
This section describes configuration that is available only in the MIB parameters of the Mediatrix unit. You can
configure these parameters as follows:



by using a MIB browser
by using the CLI
by creating a configuration script containing the configuration variables
Configuring DNS Records Randomization
You can define how the DNS A/AAAA records are accessed from the device’s internal DNS cache using the
DnsCacheRecordsRandomization variable.
The following values are available:
Table 34: DNS Cache Records Randomization Values
Value
Description
Enable
When DNS A/AAAA records are accessed from the cache, they are sent to requesting
service in a randomized order.
Disable
When DNS A/AAAA records are accessed from the cache, they are sent to requesting
service in the same order they were originally received from the network. This is the default
value.
 To configure DNS Cache records randomization:
1.
In the hocMIB, set the DnsCacheRecordsRandomization variable.
You can also use the following line in the CLI or a configuration script:
hoc.DnsCacheRecordsRandomization=”Value”
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Additional Parameters
Software Configuration Guide
where Value may be as follows:
Table 35: DNS Cache Records Randomization Values
Value Meaning
0
Disable
1
Enable
Configuring Pre-resolved Static FQDNs
You can configure up to 10 pre-resolved FQDNs. The StaticHosts table allows configuring FQDNs with static
IP addresses. When a device attempts to reach a FQDN configured in this table, the static IP addresses will
be used instead of resolving the FQDN.
The following parameters are available:
Table 36: Static Host Command Parameters
Parameter
Description
Name
Name (FQDN) of the static host. This name must be unique across the table.
The name only accepts valid FQDNs as defined by RFC 3986 (Uniform Resource Identifier
(URI): Generic Syntax). In addition, strict validation is applied, i.e. the suggested syntax
defined in RFC 1035 is enforced.
IpAddresses
List of static IP addresses associated with the FQDN specified in the StaticHosts.Name
variable.
This list contains numerical IPv4 or IPv6 addresses. IP addresses MUST be separated by
a comma (,).
Index
Index in the table. A value of zero (default) causes automatic selection of the largest
current index value + 1. If the index value already exists in the table, the insertion is
refused. This parameter is optional.
 To insert a new static host:
1.
You can use one of the following lines in the CLI or a configuration script:
hoc.InsertStaticHost Index=”value” Name="hostname" IpAddresses=”address,address1”
hoc.InsertStaticHost Name="hostname" IpAddresses=”address,address1”
where:
•
value can be an integer. This is an optional parameter.
•
hostname is a unique valid FQDN as define by RFC 3986.
•
address and address1 are numerical IPv4 or IPv6 addresses separated by a comma.
 To delete a static host:
1.
In the hocMIB, delete the host name using the Delete command.
You can also use one of the following lines in the CLI or a configuration script:
hoc.StaticHosts.Delete[Index=value]=Delete
where value can be an integer.
Updating the "sysname" or "syslocation"
You can specify the name and location of the Mediatrix unit. This information is for display purposes only and
does not affect the behavior of the unit.
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Chapter 12 - Host Parameters
Additional Parameters
 To set the sysname and syslocation parameters:
1.
In the hocMIB, set the system name in the systemName variable.
You can also use the following line in the CLI or a configuration script:
hoc.systemName="Value"
The value of this variable is also returned by the "sysName" object in SNMPv2-MIB.
2.
Set the system location in the systemLocation variable.
You can also use the following line in the CLI or a configuration script:
hoc.systemLocation="Value"
The value of this variable is also returned by the "sysLocation" object in SNMPv2-MIB.
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Reserving an IP Address
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Software Configuration Guide
H A P T E R
13
Interface Parameters
This chapter describes how to set the interfaces of the Mediatrix unit:










How to reserve an IP address in a network server.
Link Connectivity Detection
Partial Reset
Managing interfaces.
PPPoE parameters.
LLDP Configuration
Ethernet Link Configuration
DHCP Server Configuration
Ethernet Connection Speed
Configuring a MTU Value
Reserving an IP Address
Before connecting the Mediatrix unit to the network, Media5 strongly recommends that you reserve an IP
address in your network server – if you are using one – for the unit you are about to connect. This way, you
know the IP address associated with a particular unit.
Network servers generally allocate a range of IP addresses for use on a network and reserve IP addresses
for specific devices using a unique identifier for each device. The Mediatrix unit unique identifier is its media
access control (MAC) address. You can locate the MAC address as follows:



It is printed on the label located on the bottom side of the unit.
It is stored in the Device Info page of the web interface.
You can take one of the telephones connected to the Mediatrix unit and dial *#*1 on the keypad.
The MAC address of the Mediatrix unit will be stated. This applies to Mediatrix units with FXS
interfaces.
Media5 recommends to reserve an IP address with an infinite lease for each Mediatrix unit on the network.
Link Connectivity Detection
Each Ethernet port of the Mediatrix unit is associated with an Ethernet link. This information is available in the
Ethernet Ports Status section of the Network / Status page. A link has connectivity if at least one of its port
status is not disconnected.
The link connectivity is periodically polled (every 500 milliseconds). It takes two consecutive detections of the
same link state before reporting a link connectivity transition. This avoids reporting many link connectivity
transitions if the Ethernet cable is plugged and unplugged quickly.
Partial Reset
When a partial reset is triggered, the Rescue interface is configured and enabled with:

Dgw v2.0 Application
its hidden IPv4 link configuration values
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Chapter 13 - Interface Parameters


Interfaces Configuration
its hidden IPv4 address configuration
an IPv6 link-local address on all network links
Hidden values are set by the unit's profile.
Just before the Rescue is configured, all IPv4 network interfaces that could possibly conflict with the Rescue
interface are disabled.
If the BNI Service is stopped when the partial reset occurs, it is started and the above configuration is applied.
Interfaces Configuration
The Interface Configuration section allows you to add and remove up to 48 network interfaces. By default, this
section contains the following network interfaces:

The Uplink interface, which defines the uplink information required by the Mediatrix unit to
properly connect to the WAN. The Uplink network interface is the IP interface that encapsulates
the following link interface (WAN connection):
•
eth5 (Mediatrix 3000 Series models)
•
eth1 (Mediatrix 4400 Series models)
•
eth1 (Mediatrix LP/4100/C7 Series models), wan for the Mediatrix 4102S
By default, this interface uses the IPv4 DHCP connection type.

The Rescue interface, which defines the address and network mask to use to contact the
Mediatrix unit after a partial reset operation. You cannot delete this interface. Refer to
Performing a Partial Reset at http://www.media5corp.com/documentation.

The LAN interface IPv4 address and network mask.
The current status of the network interfaces is displayed in the Status page. It allows you to know which
interfaces are actually enabled. Enabled networks are activated when their configured link gets connectivity
and are deactivated as soon as the link connectivity is lost. See “Link Connectivity Detection” on page 63 for
more details.
The Interfaces Status section of the Status page displays the status of all currently enabled network interfaces,
including interfaces with an invalid configuration or waiting for a response.
When configuring network interfaces, Media5 recommends to have a syslog client properly configured and
enabled in order to receive any message related to the network interfaces behaviour. The interface used to
access the syslog client must also be properly enabled. See “Chapter 7 - Syslog Configuration” on page 35
for more details on enabling a syslog client.
Caution: Use extreme care when configuring network interfaces, especially when configuring the network
interface used to contact the unit for management. Be careful never to disable or delete the network interface
used to contact the unit. Also be careful to always set the unit’s management interface to be an interface
that you can contact.

When performing a partial reset (Refer to Performing a Partial Reset Technical Bulletin at
http://www.media5corp.com/documentation.)
Note: ), the management interface used for SNMP, CLI and WEB is automatically set to the Rescue
interface. In that case, you must change the Mediatrix unit system management network interface to
something other than Rescue. Note that you must be able to contact the interface you select.
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Dgw v2.0 Application
Interfaces Configuration
Software Configuration Guide
 To configure interfaces parameters:
1.
In the web interface, click the Network link, then the Interfaces sub-link.
Figure 22: Network – Interfaces Web Page
4
3
5
7
6
2
2.
If you want to add a new interface, enter its name in the blank field in the bottom left of the window,
then click the
button.
The name is case-sensitive. Using the special values “All”, Loop, LoopV6 and Rescue are not
allowed.
You can use the following ASCII codes in the network interface name:
49
50
51
52
53
54
55
56
57
65
66
67
68
69
70
71
72
73
74
75
76
1
2
3
4
5
6
7
8
9
A
B
C
D
E
F
G
H
I
J
K
L
77 M
78 N
79 O
80 P
81 Q
82 R
83 S
84 T
85 U
86 V
87 W
88 X
89 Y
90 Z
95 _, underscore
97 a
98 b
99 c
100 d
101 e
102 f
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
g
h
i
j
k
l
m
n
o
p
q
r
s
t
u
v
w
x
y
z
A valid network interface name must be compliant with the following rules:
•
It must start with a letter
•
It cannot contain characters other than letters, numbers, underscores.
If your Mediatrix unit contains an invalid interface name created in a previous firmware version
without the validation feature, the invalid interface name will be modified everywhere it appears on
the first restart and a syslog notification will be sent.
You can also delete an existing network interface by clicking the corresponding
cannot delete the Rescue interface.
3.
Dgw v2.0 Application
button. You
In the Interface Configuration section, select the link on which to activate the interface in the Link
column.
•
Mediatrix 3000 Series: You can select between the eth1-4 and eth5 interfaces, as well
as any defined VLANs.
•
Mediatrix LP/4100/C7 Series: You can select between the eth1 and eth2 interfaces, as
well as any defined VLANs.
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Chapter 13 - Interface Parameters
Interfaces Configuration
•
Mediatrix 4400 Series: You can select between the eth1 interface and any defined
VLANs.
A VLAN is listed with the following syntax:
Link.VLAN ID
For instance, if you have added VLAN 20 on the interface eth5, it is listed as follows:
eth5.20
Figure 23: VLAN Example
4.
Select the configuration source of the interface information in the Type drop-down menu.
Table 37: Interface Configuration Sources
Source
Description
IPv4
DHCP
The IPv4 address and network mask are provided by querying a DHCP server and
using standard DHCP fields or options. Using the DHCP configuration assumes that
you have properly set your DHCP server with the relevant information. DHCP servers
may provide a list of IP configuration parameters to use. See “DHCP Server
Configuration” on page 73 for more details.
IPv4
Static
You manually enter the IPv4 address and network mask and they remain the same
every time the Mediatrix unit restarts. Use the static configuration if you are not using a
DHCP server/PPP peer or if you want to bypass it.
IPv4
PPPoE
IPv4 over PPP connection, address and network mask are provided by the PPP peer
using IPCP. PPP peers may provide a list of IP configuration parameters to use. See
“PPPoE Configuration” on page 69 for more details.
IPv6
AutoConf
IPv6 state-less auto-configuration. See “IPv6 Autoconfiguration Interfaces” on page 67
for more details.
IPv6
Static
You manually enter the IPv6 address and network mask and they remain the same
every time the Mediatrix unit restarts. Use the IPv6 static configuration if you are not
using IPv6 stateless or stateful auto-configuration or if you want to bypass it.
Note: If no network is configured in IPv6, the unit does not have any IPv6 address, not even the Link-Local
address. When a network is configured in IPv6, the Link-Local (FE80 ::...) address is automatically created
and displayed in the Network Status information.
5.
If the interface configuration source is IPv4 Static or IPv6 Static, enter the address and network
mask (if applicable) of the network interface in the Static IP address field.
6.
If the interface configuration source is IPv4 Static or IPv6 Static, set the Static Default Router field
with the IP address of the default gateway for the network interface.
7.
Define whether or not the Mediatrix unit should attempt to activate the corresponding network
interface in the Activation drop-down menu.
It may not be possible to enable a network interface, for instance if another network interface is
already enabled in the same subnet. The actual status of network interfaces is shown in the Status
page.
8.
66
Click Apply if you do not need to set other parameters.
Dgw v2.0 Application
Interfaces Configuration
Software Configuration Guide
The current network interface information is displayed in the Status page.
Table 38: Network Interface Status
Status
Description
Disabled
The interface is not operational because it is explicitly disabled or the link
interface is unavailable.
Invalid Config
The interface is not operational because its configuration is not valid.
Network Conflict
The interface is configured with an IP address that is already used on the
network.
Link Down
The interface is configured with a link that has no connectivity.
Waiting
Response
The interface is not operational because a response from a peer or server
is required.
Active
The interface is operational.
IPv6 Autoconfiguration Interfaces
When the Type drop-down menu is set to IPv6 Auto-Conf, the network interface is an IPv6 over Ethernet
connection with IP parameters obtained by stateless auto-configuration or stateful (DHCPv6) configuration.
Autoconfiguration of IPv6 address is first initiated using state-less autoconfiguration. Stateful
autoconfiguration is initiated only if one of the following conditions is met:

The router explicitly required stateful autoconfiguration by setting the “managed” or “other” flag
of the router advertisement.

No router advertisement was received after 3 router solicitations. RFC 4861 defines the
number of router solicitations to send and the 4 seconds interval between the sent router
solicitations.
Stateless Autoconfiguration
All IPv6 addresses present in the router advertisements are applied to the network interface. Each IPv6
address is assigned a network name based on the configured network name with a suffix in the following
format: ConfiguredNetworkName-XX-Y.
XX is the address scope



GU (Global Unique)
UL (Unique Local)
LL (Link-Local)
Y is a unique ID for the address scope.
Spanning Tree Protocol vs Stateless Autoconfiguration
Many network switches use the Spanning Tree Protocol (STP) to manage Ethernet ports activity. STP uses a
detection timeout before a router advertisement is sent to the Mediatrix unit. The default value for this timeout
is usually 30 seconds. However, when the unit wants to get an IPv6 address in Stateless autoconfiguration,
this timeout is too long and the unit falls into Stateful Autoconfiguration mode before it receives the router
advertisement. This results in the unit receiving a DHCPv6 address.
To solve the issue, check if the default STP detection timeout value in your router can be modified. If so, set
it to a value of 8 s or less. If you cannot modify the timeout value, Media5 recommends to disable the Spanning
Tree Protocol on the network to which the unit is connected.
Dgw v2.0 Application
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Chapter 13 - Interface Parameters
Rescue Interface Configuration
Stateful Autoconfiguration
Stateful autoconfiguration is managed by DHCPv6. The DHCPv6 lease is negotiated according to RFC 3315
with the limitations listed in section 1.5. DHCPv6 may be used to obtain the following information (depending
on the router advertisement flags):


IPv6 addresses (when the router advertisement “managed” flag is set)
Other configuration (when the router advertisement “other” flag is set)
If only the “other” flag is set in the router advertisement, the DHCPv6 client only sends an information request
to the DHCPv6 server, otherwise it sends a DHCPv6 solicit message. If the flags change over time, only the
transitions from “not set” to “set“ are handled.
Network Interface Priority
This section describes configuration that is available only in the MIB parameters of the Mediatrix unit. You can
configure these parameters as follows:



by using a MIB browser
by using the CLI
by creating a configuration script containing the configuration variables
You can prioritize the network interfaces of the Mediatrix unit. In case of address conflicts between two or more
network interfaces, the network interface with the highest priority will remain enabled and the other interfaces
will be disabled. If the priority is the same, only the first enabled network interface will be able to use the IP
address. When a conflict ends, all network interfaces concerned automatically return to an operational state.
The actual status of network interfaces is displayed in the Status web page.
 To set the network interface priority:
1.
In the ethMIB, set the networkInterfacesPriority variable with the proper value for the
corresponding interface.
You can also use the following line in the CLI or a configuration script:
eth.networkInterfacesPriority="Value"
where Value may be any number between 0 and 100.
Rescue Interface Configuration
You can define whether or not the Mediatrix unit should attempt to activate the rescue network interface.
Caution: Please be careful when using this section.
 To enable/disable the Rescue interface:
1.
In the Rescue interface section, define whether or not the Mediatrix unit should attempt to activate
the corresponding network interface in the Activation drop-down menu.
Figure 24: Rescue Interface Configuration Section
1
It may not be possible to enable a network interface, for instance if another network interface is
already enabled in the same subnet. The actual status of network interfaces is shown in the Status
page.
2.
68
Click Apply if you do not need to set other parameters.
Dgw v2.0 Application
PPPoE Configuration
Software Configuration Guide
PPPoE Configuration
The PPPoE Configuration section applies only if you have selected the PPPoE connection type in the Interface
Configuration section of the web page.
 To configure PPPoE parameters:
1.
In the PPPoE Configuration section, set the name of the service requested to the access
concentrator (AC) when establishing the next PPPoE connection in the Service Name field.
Figure 25: PPPoE Configuration Section
1
2
3
This is used as the Service-Name field of the packet broadcasted to the access concentrators. See
RFC 2516 section 5.1 for details.
The field may be set with any string of characters, with a maximum of 255 characters.
If you leave this field empty, the Mediatrix unit looks for any access concentrator.
2.
3.
Select the authentication protocol to use for authenticating the system to the PPP peer in the
Protocol drop-down menu.
•
PAP: Use the Password Authentication Protocol.
•
CHAP: Use the Challenge Handshake Authentication Protocol.
Set the PPP user name and password that identify the system to the PPP peer during the
authentication process in the User Name and Password fields.
Caution: The User Name and Password fields are not accessible if you have the User or Observer access
right. See “Users” on page 537 for more details.
When connecting to an access concentrator, it may request that the Mediatrix unit identifies itself
with a specific user name and password.
There are no restrictions, you can use any combination of characters.
4.
Click Apply if you do not need to set other parameters.
The current PPPoE information is displayed in the Status page.
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Chapter 13 - Interface Parameters
PPPoE Configuration
PPP Negotiation
When the Mediatrix unit restarts, it establishes the connection to the access concentrator in conformance with
the RFCs listed in “PPPoE Configuration” on page 69.
When establishing a PPP connection, the Mediatrix unit goes through three distinct phases:



Discovery phase
Authentication phase
Network-layer protocol phase
Discovery Phase
The Mediatrix unit broadcasts the value of the Service Name field.
The access concentrator with a matching service name answers the Mediatrix unit.


If no access concentrator answers, this creates a “PPPoE failure” error.
If more than one access concentrators respond to the discovery, the Mediatrix unit tries to
establish the PPP connection with the first one that supports the requested service name.
Authenthication Phase
If the access concentrator requests authentication, the Mediatrix unit sends the ID/secret pair configured in
the User Name and Password fields. If the access concentrator rejects the authentication, this creates an
“authentication failure” error.
Network-Layer Protocol Phase
The Mediatrix unit negotiates an IP address. The requested IP address is the one from the last successful
PPPoE connection. If the Mediatrix unit never connected by using PPPoE (or after a factory reset), it does not
request any specific IP address.
DHCP Client Identifier Presentation
This section describes configuration that is available only in the MIB parameters of the Mediatrix unit. You can
configure these parameters as follows:



by using a MIB browser
by using the CLI
by creating a configuration script containing the configuration variables
You can define the method to use to present the value of the Client Identifier (Option 61) field through a DHCP
request. The following values are available:
Table 39: DHCP Client Identifier Presentation Parameters
Parameter
Description
Disabled
The Client Identifier option is not presented in a DHCP request.
MacAscii
The Client Identifier value is presented as the client MAC address in ASCII format. The MAC
address is represented in lowercase.
MacBinary
The Client Identifier value is presented as the client MAC address in binary format.
 To define the DHCP client identifier presentation:
1.
In the bniMIB, locate the DhcpClientGroup folder.
2.
Set the dhcpClientIdentifierPresentation variable with the proper presentation.
You can also use the following line in the CLI or a configuration script:
bni.dhcpClientIdentifierPresentation="Value"
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Dgw v2.0 Application
LLDP Configuration
Software Configuration Guide
where Value may be one of the following:
Table 40: DHCP Client Identifier Presentation Values
Value
Meaning
100
Disabled
200
MacAscii
300
MacBinary
LLDP Configuration
The Link Layer Discovery Protocol (LLDP) service is used by network devices for advertising their identity,
capabilities, and neighbors on a IEEE 802 local area network, usually wired Ethernet.
The LLDP Configuration section allows you to configure parameters related to LLDP.
 To configure LLDP parameters:
1.
In the LLDP Configuration section, set the network interface name on which LLDP should be
enabled in the Network Interface drop-down menu.
Figure 26: LLDP Configuration Section
1
2
3
LLDP cannot be activated on multiple network interfaces simultaneously.
2.
Select the address type to populate the chassis ID device identifier in the Chassis ID drop-down
menu.
Table 41: Chassis ID Parameters
Parameter
MAC Address
Description
The MAC address.
Network Address The IP address (or 0.0.0.0 if DHCP is not obtained yet).
3.
Select whether to enable the LLDP-MED protocol override of the VLAN ID, User Priority and
DiffServ values in the Override Network Policy drop-down menu.
Table 42: Override Network Policy Parameters
Parameter
Description
Enable
The service listens for LLDP advertisements, and overrides the previously
configured VLAN ID, User Priority and DiffServ with the values received.
Disable
The service only publishes its characteristics and configurations by LLDP,
and does not override anything.
The LLDP-MED (Media Endpoint Discovery) protocol is an enhancement of LLDP.
4.
Click Apply if you do not need to set other parameters.
The current LLDP information is displayed in the Status page.
Dgw v2.0 Application
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Chapter 13 - Interface Parameters
Ethernet Link Configuration
Ethernet Link Configuration
The Ethernet Link Configuration section allows you to configure the MTU as well as IEEE 802.1X
authentication.
 To configure Ethernet link parameters:
1.
In the Ethernet Link Configuration section, set the MTU field of a specific Ethernet link with a proper
value.
Figure 27: Ethernet Link Configuration Section
1
2
4
3
5
The Maximum Transmission Unit (MTU) is a parameter that determines the largest packet than can
be transmitted by an IP interface (without it needing to be broken down into smaller units). Each
interface used by TCP/IP may have a different MTU value specified. The range is from 576 to 1500
bytes. All VLAN connections use the MTU size configured on their related Ethernet link.
Note: The MTU value applied for a PPPoE connection is the smallest of the value negotiated with the server
and the value configured here.
2.
Define the IEEE 802.1x authentication protocol activation to use for a specific Ethernet link in the
corresponding 802.1x Authentication drop-down menu.
802.1X Authentication is a tag optionally added to the Ethernet frame header to specify the support
of the IEEE 802.1X Authentication. It allows getting authorization and access to secured network(s).
Table 43: 802.1x Authentication Parameters
Parameter
3.
Description
Disable
The IEEE 802.1x authentication protocol is disabled on the Ethernet link interface.
Enable
The IEEE 802.1x authentication protocol using the EAP-TLS authentication
method is enabled on the Ethernet link to get an access, through an IEEE 802.1x
EAP-TLS authenticator (such as an IEEE 802.1x capable network device), to
secured network(s). The Ethernet link interface remains always 'UP' whatever the
result of the IEEE 802.1x authentication.
Set the username used to authenticate each Ethernet link interfaces during the IEEE 802.1x EAPTLS authentication process in the corresponding EAP Username field.
This parameter is used only when the IEEE 802.1x authentication is enabled (802.1x Authentication
drop-down menu set to Enabled).
4.
Define the IEEE 802.1x level of validation used by the device to authenticate the IEEE 802.1x EAPTLS peer's certificate.
This parameter also controls the criteria used to select the host certificate sent during the
authentication handshakes.
Table 44: 802.1x Certificate Validation Parameters
72
Parameter
Description
No
Validation
No validation is performed on the peer’s certificate. Authentication with the peer is
attempted even if the system time is not synchronized. If more than one host
certificate is configured for an EAP-TLS usage, the one with the latest expiration
date is used.
Dgw v2.0 Application
EAP 802.1x Configuration
Software Configuration Guide
Table 44: 802.1x Certificate Validation Parameters (Continued)
Parameter
Trusted
And Valid
Description
Allow a connection to the network by validating if the authentication peer’s
certificate is trusted and valid. The IEEE 802.1x authentication is attempted only if
the system time is synchronized. If more than one host certificate is configured for
an EAP-TLS usage, the one that is currently valid and with the latest expiration
date is used.
5.
Indicates the configuration status of the row.
6.
Click Apply if you do not need to set other parameters.
The current status of the network interfaces is displayed in the Status page. It allows you to know
which interfaces are actually enabled.
Table 45: Ethernet Link Interface State
State
Description
Disconnected
The link interface is physically disconnected.
Up
The link interface is physically connected and considered as usable by network
interface(s).
EAP 802.1x Configuration
The EAP 802.1x Configuration section allows you to set the IEEE 802.1x version to be used by the unit.
 To configure the IEEE 802.1x version parameter:
1.
In the EAP 802.1x Configuration section, set the IEEE 802.1x version from the EAP 802.1x Version
drop-down menu.
Figure 28: EAP 802.1x Configuration Section
1
Table 46: EAP 802.1x Version Parameters
Parameter
2.
Description
Version 2001
IEEE 802.1X-2001 – Port Based Network Access Control
Version 2004
IEEE 802.1X-2004 – Port Based Network Access Control
Click Apply if you do not need to set other parameters.
DHCP Server Configuration
Note: This section applies only if you are using the DHCP connection type (“Interfaces Configuration” on
page 64).
DHCP servers generally allocate a range of IP addresses for use on a network and reserve IP addresses for
specific devices using a unique identifier for each device. The Mediatrix unit unique identifier is its media
access control (MAC) address.
Dgw v2.0 Application
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Chapter 13 - Interface Parameters
DHCP Server Configuration
You can locate the MAC address as follows:



on the label located on the bottom side of the unit.
in the System > Information web page
you can dial the following digits on a telephone connected to the Mediatrix unit:
*#*1
The Mediatrix unit answers back with its MAC address. This applies to units with FXS interfaces.
See “General POTS Configuration” on page 128 for more details.
Media5 recommends to reserve an IP address with an infinite lease for each Mediatrix unit on the network.
DHCP Negotiation
The DHCP lease is negotiated according to RFC 2131 (supports the client side of the protocol) and RFC 2132
(only sections 3.3, 3.5, 3.8 and 8.3). The following parameters are set
Table 47: DHCP Parameters
DHCP Parameter
Host Name (option 12)
Value
Set according to the Host Name parameter of the Network > Host
page (“Host Configuration” on page 53). This option cannot be empty
according to RFC 2132. If the Host Name parameter is empty, the
DHCP option 12 is not sent.
Vendor Class Identifier (option 60) Set according to the System Description parameter of the System >
Information page.
Client identifier (option 61)
74
Set according to MAC Address parameter of the System >
Information.
Dgw v2.0 Application
Ethernet Connection Speed
Software Configuration Guide
Ethernet Connection Speed
This section describes configuration that is available only in the MIB parameters of the Mediatrix unit. You can
configure these parameters as follows:



by using a MIB browser
by using the CLI
by creating a configuration script containing the configuration variables
You can set the speed and duplex of the Ethernet connection of the Mediatrix unit. The following values are
available:
Table 48: Ethernet Ports Speed and Duplex Supported
Parameter
Description
Auto
Automatic negociation of speed and duplex.
Half10
10 Mbit/s Half-duplex.
Full10
10 Mbit/s Full-duplex.
Half100
100 Mbit/s Half-duplex.
Full100
100 Mbit/s Full-duplex.
A half-duplex connection refers to a transmission using two separate channels for transmission and reception,
while a full-duplex connection refers to a transmission using the same channel for both transmission and
reception.
If unknown, set the variable to Auto so that the Mediatrix unit can automatically detect the network speed.
Caution: Whenever you force a connection speed / duplex mode, be sure that the other device and all
other intermediary nodes used in the communication between the two devices have the same configuration.
See “Speed and Duplex Detection Issues” on page 76 for more details.
The current speed and duplex configuration is displayed in the Network > Status page under the Ethernet
Ports Status section.
 To set the Ethernet connection speed and duplex:
1.
In the ethMIB, locate the portsTable folder.
2.
Set the portsSpeed variable with the proper Ethernet speed and duplex.
You can also use the following line in the CLI or a configuration script:
eth.portsSpeed="Value"
where Value may be one of the following:
Table 49: Ethernet Ports Speed and Duplex Values
Value
Dgw v2.0 Application
Meaning
100
Auto
200
Half10
300
Full10
400
Half100
500
Full100
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Chapter 13 - Interface Parameters
Ethernet Connection Speed
Speed and Duplex Detection Issues
There are two protocols for detecting the Ethernet link speed:


An older protocol called parallel detection.
A more recent protocol called auto-negotiation (IEEE 802.3u).
The auto-negotiation protocol allows to detect the connection speed and duplex mode. It exchanges
capabilities and establishes the most efficient connection. When both endpoints support the auto-negotiation,
there are no problems. However, when only one endpoint supports auto-negotiation, the parallel detection
protocol is used. This protocol can only detect the connection speed; the duplex mode cannot be detected. In
this case, the connection may not be established.
The Mediatrix unit has the possibility to force the desired Ethernet link speed and duplex mode by disabling
the auto-negotiation and selecting the proper setting. When forcing a link speed at one end, be sure that the
other end (a hub, switch, etc.) has the same configuration. To avoid any problem, the link speed and duplex
mode of the other endpoint must be exactly the same.
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Dgw v2.0 Application
Software Configuration Guide
C
H A P T E R
14
VLAN Parameters
For more details refer to the Vlan configuration technical bulletin on our documentation portal.
Dgw v2.0 Application
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Chapter 14 - VLAN Parameters
78
Dgw v2.0 Application
Introduction
C
Software Configuration Guide
H A P T E R
15
Local QoS (Quality of Service)
Configuration
This chapter describes how to configure the local QoS parameters. The local QoS tags packets sent from the
Mediatrix unit. It does not process nor classify packets coming from the network.
Introduction
QoS (Quality of Service) features enable network managers to decide on packet priority queuing. The Dgw
v2.0 application supports the Differentiated Services (DS) field and 802.1q taggings.
The Dgw v2.0 application supports the Real Time Control Protocol (RTCP), which is used to send packets to
convey feedback on quality of data delivery.
The Dgw v2.0 application does not currently support the Voice Band Data service class. It also does not
support RSVP (Resource Reservation Protocol).
Differentiated Services (DS) Field
Standards Supported
RFC 2475: An Architecture for Differentiated Services
Differentiated Services (DiffServ, or DS) is a protocol for specifying and controlling network traffic by class so
that certain types of traffic – for example, voice traffic, which requires a relatively uninterrupted flow of data,
might get precedence over other kinds of traffic.
DiffServ replaces the first bits in the ToS byte with a differentiated services code point (DSCP). It uses the
existing IPv4 Type of Service octet.
What are Differentiated Services?
Differentiated Services avoids simple priority tagging and depends on more complex policy or rule
statements to determine how to forward a given network packet. An analogy is made to travel services, in
which a person can choose among different modes of travel – train, bus, airplane – degree of comfort,
the number of stops on the route, standby status, the time of day or period of year for the trip, and so
forth.
For a given set of packet travel rules, a packet is given one of 64 possible forwarding behaviors – known
as per hop behaviors (PHBs). A six-bit field, known as the Differentiated Services Code Point (DSCP), in
the Internet Protocol header specifies the per hop behavior for a given flow of packets. The DS field
structure is presented below:
0
1
2
3
4
5
6
7
+---+---+---+---+---+---+---+---+
| DSCP
| CU
|
+---+---+---+---+---+---+---+---+
MSB
LSB
•
DSCP: Differentiated Services CodePoint.
•
CU: Currently Unused. The CU bits should always be set to 0.
For both signalling and media packets, the DSCP field is configurable independently. The entire DS field
(TOS byte) is currently configurable.
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Chapter 15 - Local QoS (Quality of Service) Configuration
Differentiated Services (DS) Field
It is the network administrator’s responsibility to provision the Mediatrix unit with standard and correct values.
 To configure the Mediatrix unit DiffServ value:
1.
In the web interface, click the Network link, then the QoS sub-link.
Figure 29: Network – QoS Web Page
2
3
2.
Set the default Differentiated Services value used by the unit for all generated packets in the Default
DiffServ (IPv4) field.
You can override this value by setting specific service class values. See “Specific Service Class
Configuration” on page 81 for more details.
This 8-bit value is directly set in the TOS field (2nd byte) of the header of transmitted IPv4 packets,
allowing you to use either DiffServ or TOS mapping.
The DiffServ value is 1 octet scalar ranging from 0 to 255. The DSCP default value should be
101110. This results in the DS field value of 10111000 (184d). This default value would result in a
value of “101” precedence bits, low delay, high throughput, and normal reliability in the legacy IP
networks (RFC 791, RFC 1812). Network managers of legacy IP networks could use the abovementioned values to define filters on their routers to take advantage of priority queuing. The default
value is based on the Expedited Forwarding PHB (RFC 2598) recommendation.
Note: RFC 3168 now defines the state in which to set the two least significant bits in the TOS byte. On the
other hand, this RFC only applies to TCP transmissions and the bits are thus set to “0” in the Mediatrix unit.
This has the following effects:
• The TOS values for UDP packets are the same as in the MIB.
• The TOS values for TCP packets are equal to the closest multiple of 4 value that is not greater than the
value in the MIB.
You can find references on DS field under the IETF working group DiffServ. For more information,
please refer to the following RFC documents:
3.
•
Definition of the Differentiated Services Field (DS Field) in the IPv4 and IPv6 Headers
(RFC 2474)
•
An Architecture for Differentiated Services (RFC 2475)
•
Assured Forwarding PHB Group (RFC 2597)
•
An Expedited Forwarding PHB (RFC 2598)
Set the Default Traffic Class value used by the unit for all generated IPv6 packets in the Default
Traffic Class (IPv6) field.
Specific service class values may be set in the Service Classes table. See “Specific Service Class
Configuration” on page 81 for more details.
The 8-bit Traffic Class field in the IPv6 header is available for use by originating nodes and/or
forwarding routers to identify and distinguish between different classes or priorities of IPv6 packets.
4.
80
Click Submit if you do not need to set other parameters.
Dgw v2.0 Application
IEEE 802.1q
Software Configuration Guide
IEEE 802.1q
The 802.1q standard recommends the use of the 802.1q VLAN tags for Ethernet frames traffic prioritization.
VLAN tags are 4-byte headers in which three bits are reserved for priority indication. The values of the priority
bits shall be provisioned.
The 802.1q standard comprises the 802.1p standard.
It is the network administrator’s responsibility to provision the Mediatrix unit with standard and correct values.
 To enable the IEEE 802.1q user priority configuration:
1.
In the Ethernet 802.1Q Tagging Configuration section of the QoS page, select Enable in the Enable
column for each interface on which you want to enable user priority tagging.
Figure 30: Ethernet 802.1Q Tagging Configuration Section
1
2
The VLAN ID part of the 802.1Q tag is always set to 0.
2.
Set the default user priority value each interface uses when tagging packets in the Default User
Priority column.
You can override each value by setting specific service class values. See “Specific Service Class
Configuration” on page 81 for more details.
The user priority is a 3 bit field in the 802.1Q tag that carries a priority value ranging from 0 to 7 and
may be used by switches to prioritize traffic. The 802.1q default priority value should be 6 for both
signalling and media packets.
3.
Click Submit if you do not need to set other parameters.
Specific Service Class Configuration
You can override the default value set in the DiffServ and 802.1q sections for each service class of the
Mediatrix unit:



Signalling
Voice
T.38
 To set specific service class values:
1.
In the Service Class Configuration section of the QoS page, set a specific DiffServ value for each
class in the DiffServ (IPv4) column.
Figure 31: Service Class Configuration Section
1
2
3
See “Differentiated Services (DS) Field” on page 79 for more details.
2.
Dgw v2.0 Application
Set the Default Traffic Class value used in IPv6 packets for each class in the Traffic Class (IPv6)
column.
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Chapter 15 - Local QoS (Quality of Service) Configuration
Network Traffic Control Configuration
The 8-bit Traffic Class field in the IPv6 header is available for use by originating nodes and/or
forwarding routers to identify and distinguish between different classes or priorities of IPv6 packets.
3.
Set a specific user priority for each class in the User Priority column.
See “IEEE 802.1q” on page 81 for more details.
4.
Click Submit if you do not need to set other parameters.
Network Traffic Control Configuration
You can apply a bandwidth limitation on the network interfaces. The limitations are applied on raw data on the
physical link and not only on the payload of the packets. All headers, checksums and control bits (TCP, IP,
CRC, etc.) are considered in the actual bandwidth.
A bandwidth limitation is applied on a physical link and not on a high-level network interfaces. All high-level
network interfaces (including VLANs) using the same physical link are affected by a configured limitation. This
limitation is applied egress only (outgoing traffic).
If the NTC service is stopped, this section is not displayed in the QoS page. See “Chapter 4 - Services” on
page 23 on information on how to start the service. Starting the NTC service enables Traffic Shaping even if
bandwidth limitation is disabled.
Bandwidth limitation is an average of the amount of data sent per second. It is thus normal that the unit sends
a small burst of data after a period of silence.
Note that the NTC service sends packets on the physical link according to their respective priorities as
described below. Lower priority packets are dropped first.
Table 50: Physical Link Priorities
Priority
Description
1
Highest priority. Packets originating from the unit with 802.1p priority set to 7.
2
Packets originating from the unit with 802.1p priority set to 6.
3
Packets originating from the unit with 802.1p priority set to 5.
4
Packets originating from the unit with 802.1p priority set to 4.
5
Packets originating from the unit with 802.1p priority set to 3.
6
Packets originating from the unit with 802.1p priority set to 2.
7
Packets originating from the unit with 802.1p priority set to 1.
8
Packets originating from the unit with 802.1p priority set to 0.
9
Lowest priority. Packets originating from another link interface (routed packets).
Packets that exceed the defined bandwidth are eventually dropped (when the buffers are exceeded). This
implies that data bursts can suffer a slight amount of packet loss. The different codecs configured and the
desired number of simultaneous channels should be taken into account when choosing a bandwidth limit to
prevent call drops, choppy voice or inconstant ptime. The NTC service can impact the execution of other
processes if the number of packets to process is too high. (High traffic and/or low limit).
 To set network traffic control parameters:
1.
In the Network Traffic Control Configuration section of the QoS page, set the corresponding Egress
Limit field with the egress bandwidth limitation for the selected link interface.
The range is from 64 to 40960 kilobits per second.
The value 0 means no bandwidth limitation and no prioritization.
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Dgw v2.0 Application
Network Traffic Control Configuration
Software Configuration Guide
This value must be set according to the upstream bandwidth limit of the network on this link. Set to
0 (disable) if the network bandwidth exceeds 40960 kbps or if it exceeds the effective limit of this
device.
Figure 32: Network Traffic Control Configuration Section
1
2.
Dgw v2.0 Application
Click Submit if you do not need to set other parameters.
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Chapter 15 - Local QoS (Quality of Service) Configuration
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Network Traffic Control Configuration
Dgw v2.0 Application
Managing the Local Firewall
C
Software Configuration Guide
H A P T E R
16
Local Firewall Configuration
This chapter describes how to configure the local firewall parameters.





Setting the default policy
Creating/editing a firewall rule
Moving a firewall rule
Deleting a firewall rule
Disabling the local firewall
Managing the Local Firewall
The local firewall allows you to dynamically create and configure rules to filter packets. The traffic is analyzed
and filtered by all the rules configured.
Note: The Mediatrix unit’s local firewall settings do not support IPv6. See “IPv4 vs. IPv6” on page 49 for
more details.
Since this is a local firewall, rules apply only to incoming packets with the unit as destination.
Incoming packets for an IP communication established by the unit are always accepted (Example : If the
Mediatrix unit sends a DNS request, the answer will be accepted).
Rules priority is determined by their position in the table.
The maximum number of rules allowed in the configuration is 20.
Caution: Enabling the local firewall and adding rules has an impact on the Mediatrix unit’s overall
performance as the firewall requires additional processing. The more rules are enabled, the more overall
performance is affected. Furthermore, Media5 recommends to use a 30 ms packetization time when the
firewall is enabled (instead of a 20 ms ptime, for instance) in order to simultaneously use all the channels
available on the unit, especially with the Mediatrix 3632 / 4404 / 4124 / LP24 models.
Partial Reset
When a partial reset is triggered and the firewall is enabled, the configuration is rolled back if it was being
modified. A new rule is then automatically applied in the firewall to allow access to the 'Rescue' interface.
However, if the firewall is disabled, the configuration is rolled back but no rule is added.
Setting the Default Policy
The default policy defines the action the Mediatrix unit must take when a packet does not match any rule.
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Chapter 16 - Local Firewall Configuration
Managing the Local Firewall
 To set the default policy:
1.
In the web interface, click the Network link, then the Local Firewall sub-link.
Figure 33: Network – Local Firewall Web Page
2
2.
In the Local Firewall Configuration section, define the Default Policy drop-down menu.
Table 51: Default Policy Parameters
Parameter
Description
Accept
Lets the packet through.
Drop
Drops the packet without any notification.
Caution: Make sure there are some rules with the Action parameter set to Accept in the local firewall
BEFORE applying changes that set the default policy to Drop. If you do not comply with this warning, you
will lose contact with the unit and a partial or factory reset will be required.
Setting the default policy to Drop or adding a rule automatically enables the local firewall. Enabling
the local firewall may have a negative impact on performance.
Creating/Editing a Firewall Rule
The web interface allows you to create a firewall rule or modify the parameters of an existing one.
 To create or edit a firewall rule:
1.
In the Local Firewall Rules section of the Local Firewall page, do one of the following:
•
If you want to add a rule before an existing entry, locate the proper row in the table and
click the
•
button of this row.
If you want to add a rule at the end of the existing rows, click the
bottom right of the section.
button at the
Figure 34: Local Firewall Rules Section
2
3
4
5
6
7
8
1
Note: When you add a new rule, edit an existing rule, or delete a rule, you can see a yellow Yes in the
Config Modified section at the top of the window. It warns you that the configuration has been modified but
not applied (i.e., the Firewall section of the Status page differs from the Local Firewall). The Local Firewall
sub-menu is a working area where you build up a local firewall configuration. While you work in this area,
the configured parameters are saved but not applied (i.e., they are not used to filter incoming packets). The
yellow Yes flag warns you that the configuration has been modified but is not applied.
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Managing the Local Firewall
Software Configuration Guide
2.
Set the current activation state for this rule in the corresponding Activation drop-down menu.
Table 52: Firewall Rule Activation State Parameters
Parameter
Description
Enable
This rule is active in the firewall.
Disable
This rule is not in the firewall.
Only enabled rules may be applied to the firewall.
3.
Enter the source address of the incoming packet in the corresponding Source Address field.
Use one of the following syntax:
Table 53: Source Address Parameters
Parameter
Description
address[/mask]
Can either be a network IP address (using /mask) or one of the host
IP addresses. The mask must be a plain number specifying the
number of binary 1s at the left side of the network mask (a mask of
24 specifies a network mask of 255.255.255.0).
networkInterfaceName
/
The value must already exist in the Interface Configuration table
(see “Interfaces Configuration” on page 64 for more details). The
interface name is case sensitive, hence it must be entered properly.
If the specified network interface is disabled or removed, the rule is
automatically disabled thus removed from the firewall. When the
network interface is enabled or added back, the rule is automatically
enabled and applied in the firewall.
Note: It is mandatory to use the suffix “/” to indicate that the network
address of this interface is used instead of the host address.
Leaving the default empty string matches any address.
4.
Enter the source port of the incoming packet in the corresponding Source Port field.
You can enter a single port or a range of ports. In the case of a range of ports, use the following
format:
port[-port]
Leaving the default empty string means that no filtering is applied on the source port, thus matching
any port.
This parameter is only effective when the Protocol drop-down menu is set to TCP or UDP (see Step
7).
5.
Enter the destination address of the incoming packet in the corresponding Destination Address
field.
Use one of the following syntax:
Table 54: Source Address Parameters
Parameter
address
Dgw v2.0 Application
Description
Must be one of the host IP addresses. Specifying a network address
is invalid since this is a local firewall.
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Chapter 16 - Local Firewall Configuration
Managing the Local Firewall
Table 54: Source Address Parameters (Continued)
Parameter
Description
networkInterfaceName
The host address of this interface is used. The value must already
exist in the Interface Configuration table (see “Interfaces
Configuration” on page 64 for more details). The interface name is
case sensitive, hence it must be entered properly.
If the specified network interface is disabled or removed, the rule is
automatically disabled thus removed from the firewall. When the
network interface is enabled or added back, the rule is automatically
enabled and applied in the firewall.
Leaving the default empty string matches any address.
6.
Enter the destination port of the incoming packet in the corresponding Destination Port field.
You can enter a single port or a range of ports. In the case of a range of ports, use the following
format:
port[-port]
Leaving the default empty string means that no filtering is applied on the destination port, thus
matching any port.
This parameter is only effective when the Protocol drop-down menu is set to TCP or UDP (see Step
7).
7.
Select the protocol of the incoming packet to filter in the corresponding Protocol drop-down menu.
Table 55: Firewal Rule Protocol Parameters
Parameter
8.
Description
All
Matches packets using any protocols.
TCP
Matches only TCP packets.
UDP
Matches only UDP packets.
ICMP
Matches only ICMP packets.
Select the action to take in the corresponding Action field.
Table 56: Firewal Rule Action Parameters
Parameter
Description
Accept
Lets the packet through.
Reject
Sends back an ICMP port unreachable in response to the matched packet. The
packet is then dropped.
Drop
Drops the packet without any notification.
Note that if a connection is already established before creating a rule that rejects it, this connection
stays active despite the rule applied.
9.
Click the Apply button to activate the enabled rules.
The current enabled rules applied are displayed in the Network > Status web page, Firewall section,
which contains the active configuration in the firewall. You can also see that the yellow Config
Modified Yes flag is cleared.
Moving a Firewall Rule
The firewall rules sequence is very important because rules priority is determined by their position in the table.
If you want the unit to try to match one rule before another one, you must put that rule first.
88
Dgw v2.0 Application
Disabling the Local Firewall
Software Configuration Guide
 To move a rule up or down:
1.
Either click the
or
arrow of the rule you want to move until the entry is properly located.
2.
Click the Apply button to update the Network > Status web page.
Deleting a Firewall Rule
You can delete a rule from the table in the web interface.
 To delete a rule entry:
1.
Click the
button of the rule you want to move.
2.
Click the Apply button to update the Network > Status web page.
Disabling the Local Firewall
When the local firewall is enabled, it has an impact on the Mediatrix unit’s overall performance as the firewall
requires CPU power. You can disable the firewall if you do not need it, thus not impacting performance.
 To disable the firewall:
Dgw v2.0 Application
1.
In the Local Firewall Configuration section, set the default policy to Accept with no rules in the local
firewall.
2.
Restart the Mediatrix unit.
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Chapter 16 - Local Firewall Configuration
90
Disabling the Local Firewall
Dgw v2.0 Application
Managing IP Routing
C
Software Configuration Guide
H A P T E R
17
IP Routing Configuration
This chapter describes how to configure the IP Routing parameters of the Mediatrix unit.





IPv4 Forwarding
Creating/editing an IP routing rule
Moving an IP routing rule
Deleting an IP routing rule
IP routing examples
Managing IP Routing
The IP Routing service allows the Mediatrix unit to perform advanced routing based on the packet’s criteria
(source IP address and source Ethernet link), which allows the packet to be forwarded to a specific network.
You can create up to four advanced IP routes.
Note: The Mediatrix unit’s IP Routing settings do not support IPv6. See “IPv4 vs. IPv6 Availability” on
page 49 for more details.
Packets matching a list of criteria should1 use advanced IP routes instead of routes present in the main routing
table of the unit.
IP Routing works together with the following services:




Network Firewall (“Chapter 18 - Network Firewall Configuration” on page 99)
NAT (“Chapter 19 - NAT Configuration” on page 105)
DHCP server (“Chapter 20 - DHCP Server Settings” on page 113)
Network Traffic Control (“Network Traffic Control Configuration” on page 82)
These services must be properly configured.
When the IP Routing service is started, IP routing is activated even if there is no configured rule (the Mediatrix
unit will forward received packets). If the IP Routing service is stopped, IP forwarding is disabled, this tab is
greyed out and the parameters are not displayed. See “Chapter 4 - Services” on page 23 on information on
how to start the service.
Caution: Enabling the IP routing service and adding rules has an impact on the Mediatrix unit’s overall
performance as IP routing requires additional processing. The more rules are enabled, the more overall
performance is affected. Furthermore, Media5 recommends to use a 30 ms packetization time when IP
routing is enabled (instead of a 20 ms ptime, for instance) in order to simultaneously use all the channels
available on the unit, especially with the Mediatrix 3632 / 4404 / 4104 models.
IPv4 Forwarding
IPv4 forwarding allows you to control the IPv4 forwarding feature and the Advanced IP Routes. When set to
Enabled, IPv4 Forwarding is enabled and the Advanced IP Routes are applied. When set to Disabled, IPv4
Forwarding is disabled and the Advanced IP Routes are not applied (the Advanced IP Routes section of the
IP Routing page is disabled).
1. A packet matching a route uses the custom routing table first and then the main routing table if no route in the custom routing table was
able to send the packet to the desired destination IP address.
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Chapter 17 - IP Routing Configuration
Managing IP Routing
 To manage IPv4 forwarding:
1.
In the web interface, click the Network link, then the IP Routing sub-link.
2.
In the IP Routing Configuration section of the IP Routing page, define whether or not IPv4
forwarding is enabled by setting the IPv4 Forwarding drop-down menu accordingly.
Figure 35: IPv4 Forwarding Configuration Section
2
3.
Click the Submit & Apply button to update the Network > Status web page.
Creating/Editing an IP Routing Rule
The web interface allows you to create a routing rule or modify the parameters of an existing one.
 To create or edit a routing rule:
1.
In the Advanced IP Routes section of the IP Routing page, do one of the following:
•
If you want to add a rule before an existing entry, locate the proper row in the table and
click the
•
button of this row.
If you want to add a rule at the end of the existing rows, click the
bottom right of the section.
button at the
Note: When you add a new rule, edit an existing rule or delete a rule, you can see a yellow Yes in the Config
Modified section at the top of the window. It warns you that the configuration has been modified but not
applied (i.e., the Advanced IP Routes section of the Status page differs from the IP Routing page). The IP
Routing sub-menu is a working area where you build up a routing configuration. While you work in this area,
the configured parameters are saved but not applied (i.e., they are not used to route packets). The yellow
Yes flag warns you that the configuration has been modified but is not applied.
Figure 36: Advanced IP Routes Section
2
3
4
5
1
2.
Set the required state for this rule in the corresponding Activation drop-down menu.
Table 57: IP Routing Rule Activation Parameters
Parameter
Description
Enable
Activates this route.
Disable
Does not activate this route.
Only enabled rules may be applied to the routing table.
3.
92
Enter the source IP address criteria an incoming packet must have to match this rule in the Source
Address field.
Dgw v2.0 Application
Managing IP Routing
Software Configuration Guide
Use the following syntax:
Table 58: Source Address Syntax
Syntax
Description
Can either be a network IP address (using /mask) or one of the
host IP addresses. The mask must be a plain number specifying
the number of binary 1s at the left side of the network mask (a
mask of 24 specifies a network mask of 255.255.255.0). For
instance:
address[/mask]
networkInterfaceName[/]
•
192.168.0.11
•
192.168.1.0/24
The value must already exist in the Interface Configuration table
(see “Interfaces Configuration” on page 64 for more details). The
interface name is case sensitive, hence it must be entered
properly.
If the specified network interface is disabled or removed, the rule
is automatically disabled thus removed from the NAT. When the
network interface is enabled or added back, the rule is
automatically enabled and applied. For instance:
•
Lan1/ (Lan1 network address)
Note: It is mandatory to use the suffix “/” to indicate that the
network address of this interface is used instead of the host
address.
When left empty, any source address matches this rule.
4.
Enter the source link criteria an incoming packet must have to match this rule in the Source Link
field.
When left empty, packets received on any link match this rule.
5.
Select the network on which the packet is forwarded in the Forward to Network drop-down menu.
6.
Click the Submit & Apply button to activate the enabled rules.
The current applied rules applied are displayed in the Network > Status web page, Advanced IP
Routes section, which contains the active configuration of the custom routing tables. You can also
see that the yellow Config Modified Yes flag is cleared.
Note: You can revert back to the configuration displayed in the Status web page at any time (including the
disabled rules) by clicking the Rollback button at the bottom of the page. All modified settings in the IP
Routing page will be lost.
Moving an IP Routing Rule
The IP routing rules sequence is very important because only one forwarding rule is applied on a packet. Rules
priority is determined by their position in the table. If you want the unit to try to match one rule before another
one, you must put that rule first.
 To move a rule up or down:
1.
Either click the
or
arrow of the rule you want to move until the entry is properly located.
2.
Click the Submit & Apply button to update the Network > Status web page.
Deleting an IP Routing Rule
You can delete a rule from the table in the web interface.
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Chapter 17 - IP Routing Configuration
Managing IP Routing
 To delete a rule entry:
1.
Click the
button of the rule you want to move.
2.
Click the Submit & Apply button to update the Network > Status web page.
Static IPv4 Routes
You can add or delete static IPv4 routes in the Mediatrix unit. A "static" route means that the route is configured
manually by the administrator. It can be configured through two different methods: through unit provisioning
or through a DHCP server (“DHCPv4 Classless Static Route Option” on page 95).
 To manage static IPv4 routes:
1.
In the Static IP Routes section of the IP Routing page, do one of the following:
•
If you want to add a route, click the
button at the bottom of the section.
•
If you want to delete an existing route, click the
move.
button of the route you want to
Figure 37: Static IP Routes Section
2
3
4
1
This section is not available if IPv4 forwarding is disabled.
2.
Specify the destination IP address criteria that an outgoing packet must have to match this route in
the corresponding Destination field.
The supported format for the destination is:
IP address[/mask]
When specifying a network as a destination, it is mandatory to use the "/" format.
The mask must be a plain number specifying the number of binary 1s at the left side of the network
mask (a mask of 24 specifies a network mask of 255.255.255.0). For instance:
3.
•
192.168.1.5 specifies an IP address as the destination.
•
192.168.1.0/24 specifies a network address as the destination.
Select the output link (interface) name in the corresponding Link drop-down menu.
When left empty, the link is selected automatically according to the information already present in
the routing table.
4.
Define the IP address of the gateway used by the route in the corresponding Gateway field.
5.
Click the Submit & Apply button to update the Network > Status web page.
The current routes available are displayed in the Network > Status web page, IPv4 Routes section.
This section identifies the entity that installed the route.
Table 59: IPv4 Routes Protocol
Protocol
94
Description
Dhcp
The route was installed dynamically by the DHCP protocol.
Static
The route was installed by the administrator of the unit.
Kernel
The route was installed by the operating system.
Other
The route was installed by another entity.
Dgw v2.0 Application
Managing IP Routing
Software Configuration Guide
DHCPv4 Classless Static Route Option
This section describes configuration that is available only in the MIB parameters of the Mediatrix unit. You can
configure these parameters as follows:



by using a MIB browser
by using the CLI
by creating a configuration script containing the configuration variables
You can define whether or not the Classless Static Route Option is enabled. Static routes can be configured
through the Classless Static Route Option for DHCPv4 (option 121) defined in RFC 3442.
If a static route to 0.0.0.0/0 is received through option 121 while a default router is also specified (see “Default
Gateway Configuration” on page 55 for more details), the route received through option 121 has priority.
The following values are available:
Table 60: DHCPv4 Classless Static Route Option Parameters
Parameter
Description
Request
The device requests the Classless Static Route Option 121.
None
Routes received from the DHCP server are ignored.
 To define whether or not the Classless Static Route Option is enabled:
1.
In the bniMIB, locate the DhcpClientGroup folder.
2.
Set the dhcpClientClasslessStaticRouteOption variable with the proper behaviour.
You can also use the following line in the CLI or a configuration script:
bni.dhcpClientClasslessStaticRouteOption="Value"
where Value may be one of the following:
Table 61: DHCPv4 Classless Static Route Option Values
Value
Meaning
100
None
200
Request
DHCPv4 User Class Route Option
Standards Supported
•
RFC 3004 -The User Class Option for DHCP
This section describes configuration that is available only in the MIB parameters of the Mediatrix unit. You can
configure these parameters as follows:



by using a MIB browser
by using the CLI
by creating a configuration script containing the configuration variables
You can define a list of user classes to enable the User Class Route Option. The list of user classes is sent
using option 77. Hexadecimal values are supported using the ‘\xXX’ format where XX is the hexadecimal
value. When the variable is empty, user class option is not sent.
 To define a list of user classes:
1.
In the bniMIB, locate the DhcpClientGroup folder.
2.
Set the dhcpClientUserClass variable with the list of user classes.
You can also use the following line in the CLI or a configuration script:
bni.dhcpClientUserClass="Value"
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Chapter 17 - IP Routing Configuration
Network Configuration Examples
where Value may be one or more user classes.
User Class items are separated by a comma and items must not be empty.
Network Configuration Examples
The following are two examples of advanced IP routing that can be accomplished with the Mediatrix unit.
Forward Packets from the Lan1 Network to the Uplink Network with NAT
1.
Create an IP routing rule so that the packets are routed (“Managing IP Routing” on page 91).
•
Source IP: Lan1/
Remove this criterion if you want to forward all packets received on the lan link.
2.
•
Source Link: lan2
•
Destination Network: Uplink
•
Click Submit & Apply.
Create a NAT rule so that the forwarded packets going on the Uplink network use the correct source
IP address (“Creating/Editing a Source NAT Rule” on page 105).
•
3.
4.
Type: SNAT
•
Source IP: Lan1/
•
Protocol: All
•
New Address: Uplink
•
Click Submit & Apply.
Create a Network Firewall rule to let established or related packets go through the unit (if the default
policy is not set to Accept) (“Managing the Network Firewall” on page 99).
•
Connection State: Established or Related
•
Action: Accept
Create a Network Firewall rule to let the packets pass from the Lan1 network to the Uplink network
(if the default policy is not set to Accept). All response packets will be accepted by the previous rule
(“Managing the Network Firewall” on page 99).
•
Source IP: Lan1/
Use additional rules or set the default policy to Accept if you want to forward packets
received on the lan link with a source address that does not match the Lan1 subnet.
•
Connection State: New
•
Action: Accept
•
Click Submit & Apply.
Configure Port Forwarding for a Web Server Located on the LAN
1.
Make sure the IP Routing service is started (to activate IP forwarding).
2.
Create a NAT rule (“Creating/Editing a Destination NAT Rule” on page 109).
This will change the destination of an HTTP packet originally destined to the Mediatrix unit with the
IP:Port of the Web server on the LAN side (to make sure the unit does not process the packet but
forwards it on the Lan1 network).
•
Type: DNat
•
Destination IP: Uplink
•
Destination Port: 8080
2. The source link name may vary depending on the unit model you have.
96
Dgw v2.0 Application
Network Configuration Examples
3.
Software Configuration Guide
•
Protocol: TCP
•
New Address: 192.168.0.11:80 (IP:Port of the Web server on the LAN side)
•
Click Submit & Apply.
Create a NAT rule (“Creating/Editing a Source NAT Rule” on page 105).
This will change the source IP address of the packet before it is sent on the Lan1 network (to make
sure the Web browser can reply correctly to the request).
4.
5.
Dgw v2.0 Application
•
Type: SNat
•
Destination IP: 192.168.0.11
•
Destination Port: 80
•
Protocol: TCP
•
New Address: Lan1
•
Click Submit & Apply.
Create a Network Firewall rule to let established or related packets go through the unit (if the default
policy is not set to Accept) (“Managing the Network Firewall” on page 99).
•
Connection State: Established or Related
•
Action: Accept
Create a Network Firewall rule to let the packets pass from the Uplink network to the Lan1 network
(if the default policy is not set to Accept). All response packets will be allowed by the previous rule
(“Managing the Network Firewall” on page 99).
•
Destination IP: 192.168.0.11
•
Destination Port: 80
•
Protocol: TCP
•
Action: Accept
•
Click Submit & Apply.
97
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98
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Managing the Network Firewall
C
Software Configuration Guide
H A P T E R
18
Network Firewall Configuration
This chapter describes how to configure the network firewall parameters.





Setting the default policy
Creating/editing a firewall rule
Moving a firewall rule
Deleting a firewall rule
Disabling the network firewall
Managing the Network Firewall
The network firewall allows dynamically creating and configuring rules to filter packets forwarded by the unit.
Since this is a network firewall, rules only apply to packets forwarded by the unit. The traffic is analyzed and
filtered by all the rules configured.
Note: The Mediatrix unit’s network firewall settings do not support IPv6. See “IPv4 vs. IPv6 Availability” on
page 49 for more details.
If no rule matches the incoming packet, the default policy is applied. A rule's priority is determined by its index
in the table.
Rules using Network Names are automatically updated as the associated IP addresses and network mask are
modified.
If the Network Firewall service is stopped, all forwarded traffic is accepted, this tab is greyed out and the
parameters are not displayed. See “Chapter 4 - Services” on page 23 on information on how to start the
service.
The maximum number of rules allowed in the configuration is 20.
Caution: Enabling the network firewall and adding rules has an impact on the Mediatrix unit’s overall
performance as the firewall requires additional processing. The more rules are enabled, the more overall
performance is affected. Furthermore, Media5 recommends to use a 30 ms packetization time when the
firewall is enabled (instead of a 20 ms ptime, for instance) in order to simultaneously use all the channels
available on the unit, especially with the Mediatrix 3632 / 4404 / 4104 models.
Setting the Default Policy
The default policy defines the action the Mediatrix unit must take when a forwarded packet does not match
any rules.
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Managing the Network Firewall
 To set the default policy:
1.
In the web interface, click the Network link, then the Network Firewall sub-link.
Figure 38: Network – Network Firewall Web Page
2
2.
In the Network Firewall Configuration section, define the default policy in the Default Policy dropdown menu.
Table 62: Default Policy Parameters
Parameter
Description
Accept
Lets the packet through.
Drop
Drops the packet without any notification.
Setting the default policy to Drop or adding a rule automatically enables the network firewall.
Enabling the network firewall may have a negative impact on performance.
Creating/Editing a Network Firewall Rule
The web interface allows you to create a network firewall rule or modify the parameters of an existing one.
 To create or edit a network firewall rule:
1.
In the Network Firewall Rules section of the Network Firewall page, do one of the following:
•
If you want to add a rule before an existing entry, locate the proper row in the table and
click the
•
button of this row.
If you want to add a rule at the end of the existing rows, click the
bottom right of the section.
button at the
Figure 39: Network Firewall Rules Section
2
3
4
5
6
7
8
9
1
Note: When you add a new rule, edit an existing rule or delete a rule, you can see a yellow Yes in the Config
Modified section at the top of the window. It warns you that the configuration has been modified but not
applied (i.e., the Firewall section of the Status page differs from the Network Firewall page). The Network
Firewall page is a working area where you build up a network firewall configuration. While you work in this
area, the configured parameters are saved but not applied (i.e., they are not used to filter packets). The
yellow Yes flag warns you that the configuration has been modified but is not applied.
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2.
Software Configuration Guide
Set the required state for this rule in the corresponding Activation drop-down menu.
Table 63: Firewall Rule Activation Parameters
Parameter
Description
Enable
This rule is active in the firewall.
Disable
This rule is not in the firewall.
Only enabled rules may be applied to the firewall.
3.
Enter the source address of the incoming packet in the corresponding Source Address or Interface
field.
Use one of the following syntax:
Table 64: Source Address Syntax
Syntax
address[/mask]
networkInterfaceName/
Description
Network IP address (using /mask). The mask must be a plain
number specifying the number of binary 1s at the left side of the
network mask (a mask of 24 specifies a network mask of
255.255.255.0). For instance:
•
192.168.0.11
•
192.168.1.0/24
The value must already exist in the Interface Configuration table
(see “Interfaces Configuration” on page 64 for more details). The
interface name is case sensitive, hence it must be entered
properly.
If the specified network interface is disabled or removed, the rule
is automatically disabled thus removed from the firewall. When
the network interface is enabled or added back, the rule is
automatically enabled and applied in the firewall. For instance:
•
Lan1/ (Lan1 network address)
Note: It is mandatory to use the suffix “/” to indicate that the
network address of this interface is used instead of the host
address.
Leaving the default empty string matches any address.
4.
Enter the source port of the incoming packet in the corresponding Source Port field.
You can enter a single port or a range of ports. This field supports the following syntax:
port[-port]
Leaving the default empty string means that no filtering is applied on the source port, thus matching
any port.
This parameter is only effective when the Protocol drop-down menu is set to TCP or UDP (see Step
7).
5.
Dgw v2.0 Application
Enter the destination address of the incoming packet in the corresponding Destination Address or
Interface field.
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Managing the Network Firewall
Use one of the following syntax:
Table 65: Source Address Syntax
Syntax
Description
Network IP address (using /mask). The mask must be a plain
number specifying the number of binary 1s at the left side of the
network mask (a mask of 24 specifies a network mask of
255.255.255.0). For instance:
address[/mask]
•
192.168.0.11
•
192.168.1.0/24
The value must already exist in the Interface Configuration table
(see “Interfaces Configuration” on page 64 for more details). The
interface name is case sensitive, hence it must be entered
properly.
networkInterfaceName/
If the specified network interface is disabled or removed, the rule
is automatically disabled thus removed from the firewall. When
the network interface is enabled or added back, the rule is
automatically enabled and applied in the firewall. For instance:
•
Lan1/ (Lan1 network address)
Note: It is mandatory to use the suffix “/” to indicate that the
network address of this interface is used instead of the host
address.
Leaving the default empty string matches any address.
6.
Enter the destination port of the incoming packet in the corresponding Destination Port field.
You can enter a single port or a range of ports. This field supports the following syntax:
port[-port]
Leaving the default empty string means that no filtering is applied on the destination port, thus
matching any port.
This parameter is only effective when the Protocol drop-down menu is set to TCP or UDP (see Step
7).
7.
Select the protocol of the incoming packet to filter in the corresponding Protocol drop-down menu.
Table 66: Firewall Rule Protocol Parameters
Parameter
8.
Description
All
Matches packets using any protocols.
TCP
Matches only TCP packets.
UDP
Matches only UDP packets.
ICMP
Matches only ICMP packets.
Set the corresponding Connection State drop-down menu with the connection state associated with
the incoming packet.
The connection state can be one of the following:
Table 67: Connection State Parameters
State
102
Description
All
Match packets in any state.
New
Match packets that are not part of an existing connection.
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Disabling the Network Firewall
Software Configuration Guide
Table 67: Connection State Parameters (Continued)
State
Description
Established Or Related Match packets that are part of an existing connection.
9.
Select the action to take in the corresponding Action field.
Table 68: Network Firewall Rule Action Parameters
Parameter
Description
Accept
Lets the packet through.
Reject
Sends back an ICMP port unreachable in response to the matched packet. The
packet is then dropped.
Drop
Drops the packet without any notification.
Note that if a connection is already established before creating a rule that rejects it, this connection
stays active despite the rule applied.
10.
Click the Apply button to activate the enabled rules.
The current enabled rules applied are displayed in the Network > Status web page, which contains
the active configuration in the network firewall. You can also see that the yellow Config Modified
Yes flag is cleared.
Note: You can revert back to the configuration displayed in the Network > Status web page at any time
(including the disabled rules) by clicking the Rollback button at the bottom of the page. All modified settings
in the Network > Network Firewall page will be lost.
Moving a Network Firewall Rule
The firewall rules sequence is very important because only one network firewall rule is applied on a packet.
Rules priority is determined by their position in the table. If you want the unit to try to match one rule before
another one, you must put that rule first.
 To move a rule up or down:
1.
Either click the
or
arrow of the rule you want to move until the entry is properly located.
2.
Click the Apply button to update the Network > Status web page.
Deleting a Network Firewall Rule
You can delete a rule from the table in the web interface.
 To delete a rule entry:
1.
Click the
button of the rule you want to move.
2.
Click the Apply button to update the Network > Status web page.
Disabling the Network Firewall
When the network firewall is enabled, it has an impact on the Mediatrix unit’s overall performance as the
firewall requires additional processing. You can disable the firewall if you do not need it, thus not impacting
performance. To disable the network firewall, you must stop the NFW service in the System > Services page.
See “Chapter 4 - Services” on page 23 for more details on how to stop a service. All forwarded traffic is allowed
when the network firewall service is stopped.
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Introduction
C
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H A P T E R
19
NAT Configuration
This chapter describes how to configure the NAT parameters of the Mediatrix unit.




Creating/editing a Source NAT
Creating/editing a Destination NAT
Moving a NAT rule
Deleting a NAT rule
Introduction
Network Address Translation (NAT, also known as network masquerading or IP masquerading) rewrites the
source and/or destination addresses/ports of IP packets as they pass through a router or firewall. It is most
commonly used to connect multiple computers to the Internet (or any other IP network) by using one IP
address. This allows home users and small businesses to cheaply and efficiently connect their network to the
Internet. The basic purpose of NAT is to multiplex traffic from the internal network and present it to the Internet
as if it was coming from a single computer having only one IP address.
The Mediatrix unit’s NAT service allows the dynamic creation and configuration of network address translation
rules. Depending on some criteria, the packet matching the rule may see its source or destination address
modified.
There are two types of NAT rules:


Source rules: They are applied on the source address of outgoing packets.
Destination rules: They are applied on the destination address of incoming packets.
A rule's priority is determined by its index in the Source NAT or Destination NAT tables.
If the NAT service is stopped, this tab is greyed out and the parameters are not displayed. See “Chapter 4 Services” on page 23 on information on how to start the service.
The maximum number of rules allowed in the configuration is 10 of each Source NAT and Destination NAT.
Caution: Adding source or destination NAT rules has an impact on the Mediatrix unit’s overall performance
as the NAT requires additional processing. The more rules are enabled, the more overall performance is
affected. Furthermore, Media5 recommends to use a 30 ms packetization time when the NAT is enabled
(instead of a 20 ms ptime, for instance) in order to simultaneously use all the channels available on the unit,
especially with the Mediatrix 3632 / 4404 /4104 models.
Partial Reset
When a partial reset is triggered, the configuration is rolled back if it was being modified.
A new rule is then automatically applied in the source and in the destination NAT tables to prevent incorrect
rules from blocking access to the unit. If those rules are not the first priority, they are raised. If there are no
rules in the tables, the new rules are not added since there are no rules to override.
Creating/Editing a Source NAT Rule
SNAT rules are executed after the routing decision, before the packet leaves the unit.
The web interface allows you to create a source NAT rule or modify the parameters of an existing one. The
following parameters must all match to apply a SNAT rule to a packet:

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Source Address
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Chapter 19 - NAT Configuration




Creating/Editing a Source NAT Rule
Source Port
Destination Address
Destination Port
Protocol
When the above parameters all match, then a new source IP address/port is applied to the packet.
 To create or edit a source NAT rule:
1.
In the web interface, click the Network link, then the NAT sub-link.
Figure 40: Source Network Address Translation Rules Section
3
4
5
6
7
8
9
2
2.
In the Source Network Address Translation Rules section of the NAT page, do one of the following:
•
If you want to add a rule before an existing entry, locate the proper row in the table and
click the
•
button of this row.
If you want to add a rule at the end of the existing rows, click the
bottom right of the section.
button at the
Note: When you add a new rule, edit an existing rule or delete a rule, you can see a yellow Yes in the Config
Modified section at the top of the window. It warns you that the configuration has been modified but not
applied (i.e., the Network Address Translation section of the Status page differs from the NAT page). The
NAT page is a working area where you build up a NAT configuration. While you work in this area, the
configured parameters are saved but not applied (i.e., they are not used in the NAT). The yellow Yes flag
warns you that the configuration has been modified but is not applied.
3.
Set the required state for this rule in the corresponding Activation drop-down menu.
Table 69: Source NAT Rule Activation Parameters
Parameter
Description
Enable
This SNAT rule is enabled.
Disable
This SNAT rule is disabled.
Only enabled rules may be applied to the Source NAT.
4.
106
Enter the source address of the incoming packet in the corresponding Source Address field.
Dgw v2.0 Application
Creating/Editing a Source NAT Rule
Software Configuration Guide
Use one of the following syntax:
Table 70: Source Address Syntax
Syntax
address[/mask]
networkInterfaceName[/]
Description
Can either be a network IP address (using /mask) or one of the
host IP addresses. The mask must be a plain number specifying
the number of binary 1s at the left side of the network mask (a
mask of 24 specifies a network mask of 255.255.255.0). For
instance:
•
192.168.0.11
•
192.168.1.0/24
The value must already exist in the Interface Configuration table
(see “Interfaces Configuration” on page 64 for more details). The
interface name is case sensitive, hence it must be entered
properly.
If the specified network interface is disabled or removed, the rule
is automatically disabled thus removed from the NAT. When the
network interface is enabled or added back, the rule is
automatically enabled and applied in the NAT. For instance:
•
Lan1 (Lan1 IP address)
•
Lan1/ (Lan1 network address)
Leaving the default empty string matches any address.
5.
Enter the source port of the incoming packet in the corresponding Source Port field.
You can enter a single port or a range of ports. This field supports the following syntax:
port[-port]
Leaving the default empty string means that no filtering is applied on the source port, thus matching
any port.
This parameter is only effective when the Protocol drop-down menu is set to TCP or UDP (see Step
7).
6.
Enter the destination address of the incoming packet in the corresponding Destination Address
field.
Use one of the following syntax:
Table 71: Destination Address Syntax
Syntax
address[/mask]
Dgw v2.0 Application
Description
Can either be a network IP address (using /mask) or one of the
host IP addresses. The mask must be a plain number specifying
the number of binary 1's at the left side of the network mask (a
mask of 24 specifies a network mask of 255.255.255.0). For
instance:
•
192.168.0.11
•
192.168.1.0/24
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Creating/Editing a Source NAT Rule
Table 71: Destination Address Syntax (Continued)
Syntax
Description
The host address of this interface is used. The value must already
exist in the Interface Configuration table (see “Interfaces
Configuration” on page 64 for more details). The interface name is
case sensitive, hence it must be entered properly.
networkInterfaceName/
If the specified network interface is disabled or removed, the rule
is automatically disabled thus removed from the NAT. When the
network interface is enabled or added back, the rule is
automatically enabled and applied in the Source NAT. For
instance:
•
Lan1/ (Lan1 network address)
Note: It is mandatory to use the suffix “/” to indicate that the
network address of this interface is used instead of the host
address.
Leaving the default empty string matches any address.
7.
Enter the destination port of the incoming packet in the corresponding Destination Port field.
You can enter a single port or a range of ports. This field supports the following format:
port[-port]
Leaving the default empty string means that no filtering is applied on the destination port, thus
matching any port.
This parameter is only effective when the Protocol drop-down menu is set to TCP or UDP (see Step
7).
8.
Select the protocol of the incoming packet to NAT in the corresponding Protocol drop-down menu.
Table 72: Source NAT Rule Protocol Parameters
Parameter
9.
Description
All
Matches packets using any protocols.
TCP
Matches only TCP packet.
UDP
Matches only UDP packets.
ICMP
Matches only ICMP packets.
Enter the new address applied to the source of the packet in the New Address field.
Use the following syntax:
Table 73: New Address Syntax
Syntax
address[:port]
10.
Description
Any IP address. When specifying a port number, it is mandatory to
have the protocol set to TCP or UDP.
Click the Apply button to activate the enabled rules.
The current enabled rules applied are displayed in the Network > Status web page, Network
Address Translation section, which contains the active configuration in the NAT. You can also see
that the yellow Config Modified Yes flag is cleared.
Note: You can revert back to the configuration displayed in the Status web page at any time (including the
disabled rules) by clicking the Rollback button at the bottom of the page. All modified settings in the NAT
page will be lost.
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Software Configuration Guide
Creating/Editing a Destination NAT Rule
The web interface allows you to create a Destination NAT rule or modify the parameters of an existing one.
This creates a rule that allows remote computers (e.g., public machines on the Internet) to connect to a specific
computer within the private LAN, depending on the port used to connect. A destination NAT is also known as
port forwarding or virtual server.
DNAT rules are executed before the routing decision, as the packet enters the unit. Therefore it is important
to configure the Network Firewall (“Chapter 18 - Network Firewall Configuration” on page 99) with respect to
the DNAT rules. An example of this would be port forwarding where the DNAT changes the routed address of
a packet to a new IP address/port. The Network Firewall must also accept connection to this IP/port in order
for the port forwarding to work.
The following parameters must all match to apply a DNAT rule to a packet:





Source Address
Source Port
Destination Address
Destination Port
Protocol
When the above parameters all match, then a new destination IP address/port is applied to the packet.
 To create or edit a Destination NAT rule:
1.
In the Destination Network Address Translation Rules section of the NAT page, do one of the
following:
•
If you want to add a rule before an existing entry, locate the proper row in the table and
click the
•
button of this row.
If you want to add a rule at the end of the existing rows, click the
bottom right of the section.
button at the
Figure 41: Destination Network Address Translation Rules Section
2
3
4
5
6
7
8
1
Note: When you add a new rule, edit an existing rule, or delete a rule, you can see a yellow Yes in the
Config Modified section at the top of the window. It warns you that the configuration has been modified but
not applied (i.e., the Network Address Translation section of the Status page differs from the NAT page).
The NAT page is a working area where you build up a NAT configuration. While you work in this area, the
configured parameters are saved but not applied (i.e., they are not used in the NAT). The yellow Yes flag
warns you that the configuration has been modified but is not applied.
2.
Set the required state for this rule in the corresponding Activation drop-down menu.
Table 74: Destination NAT Rule Activation Parameters
Parameter
Description
Enable
This DNAT rule is enabled.
Disable
This DNAT rule is disabled.
Only enabled rules may be applied to the Destination NAT.
3.
Dgw v2.0 Application
Enter the source address of the incoming packet in the corresponding Source Address field.
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Creating/Editing a Destination NAT Rule
Use one of the following syntax:
Table 75: Source Address Syntax
Syntax
address[/mask]
networkInterfaceName/
Description
Can either be a network IP address (using /mask) or one of the
host IP addresses. The mask must be a plain number specifying
the number of binary 1's at the left side of the network mask (a
mask of 24 specifies a network mask of 255.255.255.0). For
instance:
•
192.168.0.11
•
192.168.1.0/24
The host address of this interface is used. The value must already
exist in the Interface Configuration table (see “Interfaces
Configuration” on page 64 for more details). The interface name is
case sensitive, hence it must be entered properly.
If the specified network interface is disabled or removed, the rule
is automatically disabled thus removed from the NAT. When the
network interface is enabled or added back, the rule is
automatically enabled and applied in the Destination NAT. For
instance:
•
Lan1/ (Lan1 network address)
Note: It is mandatory to use the suffix “/” to indicate that the
network address of this interface is used instead of the host
address.
Leaving the default empty string matches any address.
4.
Enter the source port of the incoming packet in the corresponding Source Port field.
You can enter a single port or a range of ports. This field supports the following format:
port[-port]
Leaving the default empty string means that no filtering is applied on the source port, thus matching
any port.
This parameter is only effective when the Protocol drop-down menu is set to TCP or UDP (see Step
7).
5.
Enter the destination address of the incoming packet in the corresponding Destination Address
field.
Use one of the following syntax:
Table 76: Destination Address Syntax
Syntax
address[/mask]
110
Description
Can either be a network IP address (using /mask) or one of the
host IP addresses. The mask must be a plain number specifying
the number of binary 1's at the left side of the network mask (a
mask of 24 specifies a network mask of 255.255.255.0). For
instance:
•
192.168.0.11
•
192.168.1.0/24
Dgw v2.0 Application
Creating/Editing a Destination NAT Rule
Software Configuration Guide
Table 76: Destination Address Syntax (Continued)
Syntax
Description
networkInterfaceName[/]
The host address of this interface is used. The value must already
exist in the Interface Configuration table (see “Interfaces
Configuration” on page 64 for more details). The interface name is
case sensitive, hence it must be entered properly.
If the specified network interface is disabled or removed, the rule
is automatically disabled thus removed from the NAT. When the
network interface is enabled or added back, the rule is
automatically enabled and applied in the Destination NAT. For
instance:
•
Lan1 (Lan1 IP address)
•
Lan1/ (Lan1 network address)
Leaving the default empty string matches any address.
6.
Enter the destination port of the incoming packet in the corresponding Destination Port field.
You can enter a single port or a range of ports. This field supports the following format:
port[-port]
Leaving the default empty string means that no filtering is applied on the destination port, thus
matching any port.
This parameter is only effective when the Protocol drop-down menu is set to TCP or UDP (see Step
7).
7.
Select the protocol of the incoming packet to NAT in the corresponding Protocol drop-down menu.
Table 77: Destination NAT Rule Protocol Parameters
Parameter
8.
Description
All
Matches packets using any protocols.
TCP
Matches only TCP packets.
UDP
Matches only UDP packets.
ICMP
Matches only ICMP packets.
Enter the new address of the packet in the New Address field.
Use the following syntax:
Table 78: New Address Syntax
Syntax
address[:port]
9.
Description
Any IP address. When specifying a port number, it is mandatory to
have the protocol set to TCP or UDP.
Click the Apply button to activate the enabled rules.
The current enabled rules applied are displayed in the Network > Status web page, Network
Address Translation section, which contains the active configuration in the NAT. You can also see
that the yellow Config Modified Yes flag is cleared.
Note: You can revert back to the configuration displayed in the Network > Status web page at any time
(including the disabled rules) by clicking the Rollback button at the bottom of the page. All modified settings
in the Network > NAT page will be lost.
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Chapter 19 - NAT Configuration
Moving a NAT Rule
Moving a NAT Rule
The NAT rules sequence is very important because only one SNAT rule or one DNAT rule is applied on a
packet. Rules priority is determined by their position in the table. If you want the unit to try to match one rule
before another one, you must put that rule first.
 To move a rule up or down:
1.
Either click the
or
arrow of the rule you want to move until the entry is properly located.
2.
Click the Apply button to update the Network > Status web page.
Deleting a NAT Rule
You can delete a rule from the table in the web interface.
 To delete a rule entry:
1.
Click the
button of the rule you want to move.
2.
Click the Apply button to update the Network > Status web page.
Disabling the NAT
When the NAT is enabled, it has an impact on the Mediatrix unit’s overall performance as the NAT requires
additional processing. You can disable the NAT if you do not need it, thus not impacting performance. To
disable the NAT, you must stop the NAT service in the System > Services page. See “Chapter 4 - Services”
on page 23 for more details on how to stop a service.
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Introduction
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Software Configuration Guide
H A P T E R
20
DHCP Server Settings
This chapter describes how to configure the embedded DHCP server of the Mediatrix unit.
Introduction
The Mediatrix unit contains an embedded DHCP server that allocates IP addresses and provides leases to
the various subnets that are configured. These subnets could have PCs or other IP devices connected to the
unit’s LAN Ethernet connectors. These devices could be any combination of switches, PCs, IP phones, etc.
If the DHCP service is stopped, this tab is greyed out and the parameters are not displayed. See “Chapter 4 Services” on page 23 on information on how to start the service.
Note: The Mediatrix unit’s DHCP server settings do not support IPv6. See “IPv4 vs. IPv6” on page 49 for
more details.
Subnet Server
The DHCP server manages the hosts’ network configuration on a given subnet. Each subnet can be seen as
having a distinct DHCP server managing it, which is called a subnet server. To activate a subnet server for a
given network interface, the name of that network interface and the name of the subnet configuration must
match (the names are case sensitive). Only one subnet can be defined per network interface. The network
interface can be a physical interface or a logical interface (e.g., sub-interface using VLAN).
Leases
In order to assign leases, the subnet server draws from an IP address pool (or subnet scope) defined by a
start address and an end address. The subnet mask assigned to hosts is taken directly from the network
interface. All hosts on the same subnet share the same configuration. The maximum number of hosts
supported on a subnet is 254.
You can reserve IP addresses for specific hosts that are designated by their MAC address. Those addresses
are then removed from the pool of IP addresses that can be leased. Once a lease is assigned, it is removed
from the pool of IP addresses that can be leased for as long as the host keeps it.
Configuration Parameters
When an address is leased to a host, several network configuration parameters are sent to that host at the
same time according to the options found in the DHCP request. You can modify the configuration source of a
parameter. The following are the possible configuration sources:
Table 79: Parameter Configuration Sources
Source
Description
Static
The parameter is defined as a static parameter locally.
Automatic
The parameter is obtained from the network configured in the Automatic Configuration
Interface drop-down menu of this subnet (“DHCP Basic Configuration” on page 115).
Host Configuration The parameter is obtained from the host configuration.
Host Interface
Dgw v2.0 Application
The parameter is obtained from the network interface matching the subnet.
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Introduction
The following table lists the configuration parameters and their available configuration sources:
Table 80: Optional Parameter and Possible Configuration Sources
Configuration Sources
Parameter Name
Static
Automatic
Host Config
Host Interface
Domain Name
Lease time
Default gateway
List of DNS servers
List of NTP servers
List of NBNS servers
Default vs. Specific Configurations
You can use two types of configuration:


Default configurations that apply to all the subnets of the Mediatrix unit.
Specific configurations that override the default configurations.
You can define specific configurations for each subnet in your Mediatrix unit. For instance, you
could define a lease time for all the subnets of the Mediatrix unit and use the specific configuration
parameters to set a different value for one specific subnet.
The parameters available differ according to the subnet you have selected. The Default subnet has less
parameters than the specific subnets available on the Mediatrix unit.
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DHCP Basic Configuration
Software Configuration Guide
DHCP Basic Configuration
The basic configuration parameters are available only on the specific subnets configuration.
 To set the DHCP server basic parameters:
1.
In the web interface, click the Network link, then the DHCP Server sub-link.
2.
Select a specific subnet in the Select Subnet drop-down menu at the top of the window.
You have the choice between Default (applies to all subnets) and specific subnets.
3.
In the DHCP Server Configuration section of the DHCP Server page, enable the DHCP server by
selecting Enable in the DHCP Server Enable drop-down menu.
Figure 42: DHCP Server Configuration – General Parameters
3
4
5
4.
Set the start and end IP addresses of the subnet range in the Start IP Address and End IP Address
fields.
These are the addresses that the DHCP server offers to the subnets of the Mediatrix unit. The
Mediatrix unit can offer up to 254 addresses. These addresses must be within the network
interface’s subnet or the subnet server will have an invalid configuration status.
5.
Set the Automatic Configuration Interface drop-down menu with the network interface that provides
the automatic configuration (e.g.: DNS servers, NTP server, etc.) to all parameters of this subnet
that use the "Automatic" configuration source.
Note:
If you created Network Interfaces in the Interface Configuration table (Network > Interfaces) and wish to use them as the Automatic
Configuration Interface, you must first make the Network interface available to the DHCP service. Refer to “To have a new network
interface available in the DHCP service:” on page 115
6.
Click Submit if you do not need to set other parameters.
 To have a new network interface available in the DHCP service:
In the CLI or UME, type the dhcp.AddSubnet command, where Network =<Interface Name>.
Lease Time (Option 51)
The Mediatrix unit DHCP server offers a lease time to its subnets. You can use a default lease time for all
subnets or define a lease time specific to one or more subnets.
 To set the DHCP server lease time parameters:
1.
In the Lease Time (Option 51) sub-section of the DHCP Server Configuration section, define
whether or not you want to override the lease time set in the Default configuration in the Subnet
Specific drop-down menu.
This menu is available only in the specific subnets configuration.
Figure 43: DHCP Server Configuration – Lease Time Option
1
2
2.
Dgw v2.0 Application
Define the lease time (in seconds) given by the Mediatrix unit DHCP server in the Lease Time field.
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Chapter 20 - DHCP Server Settings
3.
Domain Name (Option 15)
Click Submit if you do not need to set other parameters.
Domain Name (Option 15)
The Mediatrix unit DHCP server offers a domain name to its subnets. You can use a default domain name for
all subnets or define a domain name specific to one or more subnets.
 To set the DHCP server domain name parameters:
1.
In the Domain Name (Option 15) sub-section of the DHCP Server Configuration section, enable the
domain name (option 15) by selecting Enable in the Enable Option drop-down menu.
This menu is available only in the specific subnets configuration.
Figure 44: DHCP Server Configuration – Domain Name Option
1
2
3
4
2.
Define whether or not you want to override the domain name parameters set in the Default
configuration in the Subnet Specific Value drop-down menu.
This menu is available only in the specific subnets configuration.
3.
If the domain name option is enabled, select the configuration source of the domain name
information in the Configuration Source drop-down menu.
Table 81: Domain Name Configuration Sources
Source
Description
Host
Configuration
The domain name is the one used by the unit.
Static
You manually enter a domain name.
Static Configuration Source Only
4.
If the configuration source is Static, enter the static default domain name for all subnets in the
Domain Name field.
5.
Click Submit if you do not need to set other parameters.
Default Gateway (Option 3)
The Mediatrix unit DHCP server offers a default gateway (also called default router) to its subnets.
Note: The default gateway parameters are not available in the Default interface. You must access the
specific subnets configuration to set its parameters.
116
Dgw v2.0 Application
DNS (Option 6)
Software Configuration Guide
 To set the DHCP server default gateway parameters:
1.
In the Default Gateway (Option 3) sub-section of the DHCP Server Configuration section, enable
the default gateway (option 3) by selecting Enable in the Enable Option drop-down menu
Figure 45: DHCP Server Configuration – Default Gateway Option
1
2
3
2.
Select the configuration source of the default gateway information in the Configuration Source dropdown menu.
Table 82: Default Gateway Configuration Sources
Source
Description
Host Interface
The default gateway is the host address within the client's subnet.
Static
You manually enter the value.
Static Configuration Source Only
3.
If the configuration source is Static, enter the default gateway host name or IP address of the
subnet in the Default Gateway field.
4.
Click Submit if you do not need to set other parameters.
DNS (Option 6)
The Mediatrix unit DHCP server offers up to four DNS addresses to its subnets. You can use the default DNS
addresses for all subnets or define static DNS addresses specific to one or more subnets.
 To set the DHCP server DNS parameters:
1.
In the DNS (Option 6) sub-section of the DHCP Server Configuration section, enable the DNS
servers (option 6) by selecting Enable in the Enable Option drop-down menu
This menu is available only in the specific subnets configuration.
Figure 46: DHCP Server Configuration – DNS Option
1
2
3
4
2.
Define whether or not you want to override the default values in the Subnet Specific drop-down
menu.
This menu is available only in the specific subnets configuration.
3.
Select the configuration source of the DNS information in the Configuration Source drop-down
menu.
Table 83: DNS Configuration Sources
Source
Host
Configuration
Dgw v2.0 Application
Description
The DNS servers are obtained from the host configuration.
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Chapter 20 - DHCP Server Settings
NTP (Option 42)
Table 83: DNS Configuration Sources (Continued)
Source
Description
Automatic
The DNS servers are automatically obtained from the network configured in the
Automatic Configuration Interface drop-down menu of this subnet (“DHCP
Basic Configuration” on page 115).
Static
You manually enter the value.
Static Configuration Source Only
4.
5.
If the configuration source is Static, enter the static addresses of up to four DNS servers in the
following fields:
•
Primary DNS
•
Secondary DNS
•
Third DNS
•
Fourth DNS
Click Submit if you do not need to set other parameters.
NTP (Option 42)
The Mediatrix unit DHCP server offers the addresses of up to four NTP (Network Time Protocol) servers to its
subnets. You can use the default NTP addresses for all subnets or define static DNS addresses specific to
one or more subnets.
 To set the DHCP server NTP parameters:
1.
In the NTP (Option 42) sub-section of the DHCP Server Configuration section, enable the NTP
servers (option 42) by selecting Enable in the Enable Option drop-down menu
This menu is available only in the specific subnets configuration.
Figure 47: DHCP Server Configuration – NTP Option
1
2
3
4
2.
Define whether or not you want to override the default values in the Subnet Specific drop-down
menu.
This menu is available only in the specific subnets configuration.
3.
Select the configuration source of the NTP information in the Configuration Source drop-down
menu.
Table 84: NTP Configuration Sources
Source
118
Description
Host
Configuration
The NTP servers are obtained from the host configuration.
Automatic
The NTP servers are automatically obtained from the network configured in the
Automatic Configuration Interface drop-down menu of this subnet (“DHCP
Basic Configuration” on page 115).
Dgw v2.0 Application
NBNS (Option 44)
Software Configuration Guide
Table 84: NTP Configuration Sources (Continued)
Source
Static
Description
You manually enter the value.
Static Configuration Source Only
4.
5.
If the configuration source is Static, enter the static addresses of up to four NTP servers in the
following fields:
•
Primary NTP
•
Secondary NTP
•
Third NTP
•
Fourth NTP
Click Submit if you do not need to set other parameters.
NBNS (Option 44)
The NetBIOS Name Server (NBNS) protocol, part of the NetBIOS over TCP/IP (NBT) family of protocols, is
implemented in Windows systems as the Windows Internet Name Service (WINS). By design, NBNS allows
network peers to assist in managing name conflicts.
The Mediatrix unit DHCP server offers up to four NBNS addresses to its subnets. You can use the default
NBNS addresses for all subnets or define static NBNS addresses specific to one or more subnets.
 To set the DHCP server NBNS parameters:
1.
In the NBNS (Option 44) sub-section of the DHCP Server Configuration section, enable the NBNS
servers (option 44) by selecting Enable in the Enable Option drop-down menu
This menu is available only in the specific subnets configuration.
Figure 48: DHCP Server – NBNS Option
1
2
3
2.
Define whether or not you want to override the default values in the Subnet Specific drop-down
menu.
This menu is available only in the specific subnets configuration.
3.
4.
Dgw v2.0 Application
Enter the static addresses of up to four NBNS servers in the following fields:
•
Primary NBNS
•
Secondary NBNS
•
Third NBNS
•
Fourth NBNS
Click Submit if you do not need to set other parameters.
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Chapter 20 - DHCP Server Settings
DHCP Static Leases Configuration
DHCP Static Leases Configuration
The embedded DHCP server leases addresses to the hosts that request it. The address is assigned to a host
for a configurable amount of time (as defined in “Lease Time (Option 51)” on page 115). The DHCP server
can service all subnets on which it is enabled.
 To define DHCP leases offered by the Mediatrix unit:
1.
In the web interface, click the System link, then the DHCP Leases sub-link.
2.
In the Static Leases Configuration section, if applicable, delete an existing reserved IP address by
selecting Delete in the Action drop-down next to an existing lease.
3.
If applicable, add a new lease by entering the MAC address of the device and the IP address you
want to reserve for it, then click Submit.
The static IP address is added to the Static Leases Configuration section, but not to the Current
Leases section.
4.
120
Click Submit if you do not need to set other parameters.
Dgw v2.0 Application
SBC Parameters
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Software Configuration Guide
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H A P T E R
21
SBC Configuration
This chapter describes how to configure the Sbc (Session Border Controller) service, which allows the
administrator to perform SIP to SIP normalization, call routing, NAT traversal and survivability.
Note: This web page is available only on the following models:
• Sentinel and Mediatrix 3000
For complete details on the Sbc service, refer to the Sbc User Guide at http://www.media5corp.com/
documentation
Dgw v2.0 Application
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Chapter 21 - SBC Configuration
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POTS Parameters
Page Left Intentionally Blank
POTS Status
C
Software Configuration Guide
H A P T E R
22
POTS Configuration
This chapter describes how to configure the POTS (Plain Old Telephony System) line service, which allows
you to configure the analog specification of each line, as well as gateways-specific parameters.
Note: This web page is available only on the following models:
• Mediatrix 3208 / 3216
• Mediatrix 3308 / 3316
• Mediatrix 3716
• Mediatrix 3731
• Mediatrix 3732
• Mediatrix 3741
• Mediatrix 3742
• Mediatrix 4100 Series
• Mediatrix LP Series
• Mediatrix C7 Series
POTS Status
The POTS parameters are displayed in the POTS / Status page.
Line Status
The Line Status table lists the link state of the FXS lines.
Figure 49: POTS – Status Web Page
The State column may have one of the following values:





Dgw v2.0 Application
Idle: The line is available
In Use: The line is currently used
Disabled: The line is disabled
Bypass: The line is on bypass
Down: The power of the line is down
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Chapter 22 - POTS Configuration
General POTS Configuration
FXO Line Status
Note: This web page is available only on the following models:
• Mediatrix 3208 / 3216
• Mediatrix 3308 / 3316
• Mediatrix 3716
• Mediatrix 3731
• Mediatrix 3732
• Mediatrix 3741
• Mediatrix 3742Mediatrix C730 / C733 / C731
The FXO Line Status table lists the link state updated by the link state verification mechanism.
Figure 50: POTS – FXO Line Status Table
You can enable the line state verification in “FXO Custom Basic Parameters” on page 142.
The unit does not automatically detect when a previously connected port has changed to the disconnected
status. The unit only detects the change of status when it attempts to use the port or after a restart.
The unit automatically detects within seconds when a disconnected port becomes connected.



Unknown: The line fault detection is disabled.
Up: When last polled, the line was connected.
Down: When last polled, the line was disconnected.
General POTS Configuration
The General Configuration section allows you to select the detection/generation method of caller ID.
 To configure the general POTS parameters:
1.
In the web interface, click the POTS link, then the Config sub-link.
Figure 51: POTS Web Page
2
4
2.
3
Select the detection/generation method of caller ID in the Caller ID customization drop-down menu.
This allows selecting the detection/generation method of caller ID. See “Caller ID Information” on
page 130 for more details.
Table 85: Caller ID Parameters
Parameter
Country
128
Description
Uses the default caller ID of the country defined in the Country section of the
Telephony > Misc page (“Country Configuration” on page 419).
Dgw v2.0 Application
General POTS Configuration
Software Configuration Guide
Table 85: Caller ID Parameters (Continued)
Parameter
3.
Description
EtsiDtmf
ETSI 300 659-1 (DTMF string sent between the first and second ring).
EtsiFsk
ETSI 300 659-1 (FSK (V.23) sent between the first and second ring).
Select the caller ID transmission method in the Caller ID Transmission drop-down menu.
It allows selecting the transmission type of the caller ID.
Table 86: Caller ID Transmission Parameters
Parameter
Description
Country
Uses the default caller ID of the country defined in the Country section of the
Telephony > Misc page (“Country Configuration” on page 419).
First Ring
The caller ID is sent after the first ring.
Ring Pulse
The caller ID is sent between a brief ring pulse and the first ring.
Line Reversal
Ring Pulse
The caller ID is sent between a brief ring pulse and the first ring on an inverted
polarity line.
DT-AS
The caller ID is sent after the dual tone alerting state tone.
Line Reversal
DT-AS
The caller ID is sent after the dual tone alerting state tone on an inverted
polarity line.
No Ring Pulse The caller ID is sent before the first ring.
4.
Determine the type of vocal information that can be obtained by dialing a pre-defined digit map in
the Vocal Unit Information drop-down menu.
When entering special characters on your telephone pad, the Mediatrix unit talks back to you with
relevant information.
Table 87: Caller ID Parameters
Parameter
Description
None
The vocal information feature is disabled.
All
Enable all vocal information digit maps.
To access the vocal unit information:
a.
Take one of the telephones connected to the Mediatrix unit.
b.
Dial one of the digits sequence on the keypad.
Table 88: Vocal Unit Information
Digits to Dial
5.
Dgw v2.0 Application
Information Vocally Sent by the Mediatrix unit
*#*0
List of IP addresses of the Mediatrix unit (static or
DHCP).
*#*1
MAC address of the Mediatrix unit.
*#*8
Firmware version number of the Mediatrix unit.
Click Submit if you do not need to set other parameters.
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Chapter 22 - POTS Configuration
General POTS Configuration
Caller ID Information
The caller ID is a generic name for the service provided by telephone utilities that supply information such as
the telephone number or the name of the calling party to the called subscriber at the start of a call. In call
waiting, the caller ID service supplies information about a second incoming caller to a subscriber already busy
with a phone call. However, note that caller ID on call waiting is not supported by all caller ID-capable
telephone displays.
In typical caller ID systems, the coded calling number information is sent from the central exchange to the
called telephone. This information can be shown on a display of the subscriber telephone set. In this case, the
caller ID information is usually displayed before the subscriber decides to answer the incoming call. If the line
is connected to a computer, caller information can be used to search in databases and additional services can
be offered.
The following basic caller ID features are supported:




Date and Time
Calling Line Identity
Calling Party Name
Visual Indicator (MWI)
Caller ID Generation
There are two methods used for sending caller ID information depending on the application and countryspecific requirements:


caller ID generation using DTMF signalling
caller ID generation using Frequency Shift Keying (FSK)
Note: The Dgw v2.0 Application does not support ASCII special characters higher than 127.
The displayed caller ID for all countries may be up to 20 digits for numbers and 50 digits for names.
DTMF Signalling
The data transmission using DTMF signalling is performed during or before ringing depending on the country
settings or endpoint configuration. The Mediatrix unit provides the calling line identity according to the following
standards:

Europe: ETSI 300 659-1 January 2001 (Annex B): Access and Terminals (AT); Analogue
access to the Public Switched Telephone Network (PSTN); Subscriber line protocol over the
local loop for display (and related) services; Part 1: On-hook data transmission.
FSK Generation
Different countries use different standards to send caller ID information. The Mediatrix unit is compatible with
the following widely used standards:

ETSI 300 659-1
Note: The compatibility of the Mediatrix unit is not limited to the above caller ID standards.
Continuous phase binary FSK modulation is used for coding that is compatible with:


130
BELL 202
ITU-T V.23
Dgw v2.0 Application
FXS Configuration
Software Configuration Guide
FXS Configuration
The FXS Configuration section allows you to define how a FXS endpoint behaves in certain conditions.
 To configure the FXS parameters:
1.
In the web interface, click the POTS link, then the FXS Config sub-link.
Figure 52: FXS Config Web Page
2
4
6
3
5
7
8
2.
In the FXS Configuration section, set the Line Supervision Mode drop-down menu with the power
drop and line polarity used to signal the state of a line.
Power drop and polarity reversal are also called battery drop and battery reversal.
Table 89: Line Supervision Mode Parameters
Parameter
Description
None
Power drop or polarity reversal is not used to signal the state of the
line.
DropOnDisconnect
Activates the Power Drop on Disconnect feature. A short power drop
is made at the end of a call when the call is disconnected by the
remote party.
The drop duration can be configured in the FXS Power Drop on
Disconnect Duration field (Step 5).
ReversalOnIdle
Activates the Polarity Reversal on Idle feature. The polarity of the line
is initially in reversed state. The polarity of the line returns to the
positive state when the user seizes the line or when the line rings for
an incoming call. The polarity of the line is reversed again when the
call is disconnected.
ReversalOnEstablished Activates the Polarity Reversal on Established option. The polarity of
the line is initially in the positive state. The polarity of the line is
reversed when the call is established and returns to the positive state
when the call is disconnected.
3.
Set the Disconnect Delay field with the value used to determine whether or not call clearing occurs
as soon as the called user is the first to hang up a received call.
This parameter has no effect when you are acting as the calling party.
If you set the value to 0, the call is disconnected as soon as the called user hangs up the call.
If the value is greater than 0, that value is the amount of time, in seconds, the unit waits after the
called user hangs up before signalling the end of the call.
4.
Dgw v2.0 Application
Set the Auto Cancel Timeout field with the time, in seconds, the endpoint rings before the call is
automatically cancelled.
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Chapter 22 - POTS Configuration
FXS Configuration
Setting this variable to 0 disables the timeout. Calls will not be automatically cancelled and will ring
until the party answers.
5.
Set the Inband Ringback drop-down menu to define whether or not the FXS endpoint needs to
generate a ringback for incoming ringing call.
Table 90: Inband Ringback Parameters
Parameter
6.
Description
Disable
The FXS endpoint does not play local ringback to the remote party.
Enable
The FXS endpoint plays local ringback to the remote party via the negotiated
media stream. The local ringback is generated only when the telephone is onhook. The FXS ports never play the local ringback for the call waiting.
Set the Shutdown Behavior drop-down menu with the FXS endpoint behavior when it becomes shut
down.
Table 91: FXS Shutdown Behavior Parameters
Parameter
Description
Disabled
Tone
A disabled tone is played when the user picks up the telephone and the FXS
endpoint is shut down.
Power
Drop
The loop current is interrupted when the FXS endpoint is shut down and no tone is
played when the user picks up the telephone.
A FXS endpoint becomes shut down when the operational state of the endpoint becomes Disabled
and the Shutdown Endpoint When Operational State is 'Disable' And Its Usage State Is 'idleunusable' parameter of the SIP > Endpoints page is set to Enable. See “Administration” on page 32
for more details.
This parameter is not used by FXS endpoints used for bypass when the Activation column of the
FXS Bypass section is set to Endpoint Disabled. See “FXS Bypass” on page 135 for more details.
7.
Set the Power Drop on Disconnect Duration field with the power drop duration, in milliseconds, that
is made at the end of a call when the call is disconnected by the remote party.
This value only has an effect when the Line Supervision Mode drop-down menu is set to
DropOnDisconnect.
8.
Set the Service Activation drop-down menu with the method used by the user to activate
supplementary services such as call hold, second call, call waiting, call transfer and conference call.
Table 92: Service Activation Parameters
Parameter
Flash Hook
132
Description
Service activation is performed by flash hook or hanging up.
Dgw v2.0 Application
FXS Country Customization
Software Configuration Guide
Table 92: Service Activation Parameters (Continued)
Parameter
Description
Flash Hook And
Digit
Service activation is performed by flash hook, flash hook followed by a digit
or hanging up.
The digit dialed has a different behaviour depending on the current call
context:
•
One call active with one waiting call or one call on hold:
Flash hook then dial the digit 1 or hang up: Terminate the active call
and switch to the call on hold/waiting.
Flash hook then dial the digit 2: Hold the active call and switch to the
call on hold/waiting.
Flash hook then dial the digit 3: Enter in three-party conference mode.
Flash hook then dial the digit 4: Transfer the call on hold to the active
call (not available on call waiting).
When hanging up in this context, the telephone rings to notify the user
there is still a call on hold.
•
In three-party conference mode:
Flash hook then dial the digit 1: Exit from three-party conference
mode. The third party remains active and the second party call is
terminated.
Flash hook then dial the digit 2: Exit from three-party conference
mode. The second party remains active and the third party call is
placed on hold.
When hanging up in this context, all calls are finished.
9.
Click Submit if you do not need to set other parameters.
FXS Country Customization
The FXS Country Customization section allows you to override the current default country parameters of
certain features. Refer to “Appendix A - Country-Specific Parameters” on page 651 for the pre-defined values
for a specific country.
 To define the FXS country customization parameters:
1.
In the FXS Country Configuration section, select whether or not you want to override the current
country parameters in the Override Country Customization drop down menu.
This allows overriding FXS related default country settings for the loop current and flash hook
detection features.
Figure 53: FXS Country Customization Section
1
3
2
Table 93: Line Supervision Mode Parameters
Parameter
Disable
Dgw v2.0 Application
Description
The line uses the default country FXS settings.
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Chapter 22 - POTS Configuration
FXS Country Customization
Table 93: Line Supervision Mode Parameters (Continued)
Parameter
Enable
2.
Description
The line uses the FXS country configuration set in the following
steps.
Set the Country Override Loop Current field with the loop current generated by the FXS port in ma.
When a remote end-user goes on-hook, the Mediatrix unit signals the far end disconnect by
performing a current loop drop (< 1 mA) on the analog line. This current loop drop, also referred to
as “Power Denial” mode, is typically used for disconnect supervision on analog lines. The Mediatrix
unit maintains a current drop for one second (this value cannot be configured), then a busy tone is
generated to indicate the user to hang up. See the description for the FXS Line Supervision Mode
drop-down menu in “FXS Configuration” on page 131 for more details.
When one of its analog lines goes off-hook, the Mediatrix unit controls the endpoint in a fixed loop
current mode. When selecting a country (see “Country Configuration” on page 419 for more details),
each country has a default loop current value. However, you can override this value and define your
own loop current.
Note that the actual measured current may be different than the value you set, because it varies
depending on the DC impedance. Mediatrix LP16/LP24 models: The values available for these
models are between 20 mA and 25 mA. The default value is 22 mA. If you set a value higher than
25 mA, the unit will limit the current to 25 mA.
3.
Set the Country Override Flash Hook Detection Range field.
This is the range in which the hook switch must remain pressed to perform a flash hook.
When selecting a country (see “Country Configuration” on page 419 for more details), each country
has a default minimum and maximum time value. However, you can override these values and
define your own minimum and maximum time within which pressing and releasing the plunger is
actually considered a flash hook.
The range consists of the minimal delay and maximal delay, in ms, separated by a “-”. The minimal
value allowed is 10 ms and the maximum value allowed is 1200 ms. The space character is not
allowed.
Flash hook can be described as quickly depressing and releasing the plunger in or the actual
handset-cradle to create a signal indicating a change in the current telephone session. Services
such as picking up a call waiting, second call, call on hold, and conference are triggered by the use
of the flash hook.
A flash hook is detected when the hook switch is pressed for a shorter time than would be required
to be interpreted as a hang-up.
Using the “flash” button that is present on many standard telephone handsets can also trigger a
flash hook.
4.
Click Submit if you do not need to set other parameters.
Calling Party Name of the Caller ID
1.
In the potsMIB, specify the Calling Party Name of the caller ID (CLIP) when the calling party is
tagged as private in the FxsCallerIdPrivateCallingPartyName variable.
You can also use the following line in the CLI or a configuration script:
pots.FxsCallerIdPrivateCallingPartyName="Value"
Value may be any string of characters up to 50 characters.
134
•
When empty, no Calling Party Name parameter is sent.
•
When set to 'P', no Calling Party Name parameter is sent but a Reason for Absence or
Caller Party Name parameter is sent with the value 0x50 (Private).
Dgw v2.0 Application
FXS Bypass
Software Configuration Guide
FXS Bypass
Note: This web page is available only on the following models:
• Mediatrix 3308 / 3316
• Mediatrix 3716
• Mediatrix 3731
• Mediatrix 3741
• Mediatrix 4100 Series (except Mediatrix 4102S)
The FXS Bypass section allows you to define whether or not the bypass feature is enabled. This feature allows
its users to maintain telephone services in the event of a power outage or network failure.
During normal operation, the SCN line connected to the Bypass connector is switched out of the circuit through
commuting relays. The Bypass connector can be activated by three different conditions:



When power is removed from the Mediatrix unit.
When the FXS endpoint’s operational state is disabled and the endpoint is not in use.
When the user signals the configured DTMF map.
If one of these conditions is met, a phone/fax used on a FXS connector that supports the bypass feature is
directly connected to the SCN Bypass line. The FXS connector stays in Bypass connection until:


The error conditions have been cleared.
The device connected to it is on-hook and a delay has elapsed.
Note: For the Mediatrix 3308, Mediatrix 3316, Mediatrix 3731 and Mediatrix 3741 models: If an event that
activates the FXS bypass occurs and the FXO port is in use, the bypass activation waits until the FXO port
is no longer in use. See “FXO Configuration” on page 139 for more details.
For more information on your model’s bypass feature and which FXS connectors are available for bypass,
please refer to your model’s Hardware Configuration Guide.
 To define the FXS bypass parameters:
1.
In the FXS Bypass section, specify when the bypass needs to be activated in the corresponding
Activation column’s drop-down menu.
Figure 54: FXS Bypass Section
1
2
3
Table 94: Activation Parameters
Parameter
Description
Power Off
The bypass is activated only when the unit power is off or the
POTS service is not started.
Endpoint Disabled
The bypass is activated when:
•
the operational state of the endpoint is Disable, and
•
the Shutdown Endpoint When Operational State is
'Disable' And Its Usage State Is 'idle-unusable'
parameter of the SIP > Endpoints page is set to
Enable. See “Administration” on page 32 for more
details.
The bypass is also activated for the same conditions as the ones
defined in Power Off.
Dgw v2.0 Application
135
Chapter 22 - POTS Configuration
FXS Emergency Call Override
Table 94: Activation Parameters (Continued)
Parameter
On Demand
Description
The bypass is activated when the user enters the DTMF map
configured in the corresponding Activation DTMF Map field (Step
2).
The bypass is deactivated when the user on hooks the phone for
at least the number of time configured in the corresponding
Deactivation Timeout field (Step 3).
The bypass is also activated for the same conditions as the ones
defined in Power Off.
2.
Set the corresponding Activation DTMF Map field with the DTMFs to signal to enable the bypass.
This field is only used when the corresponding Activation drop-down menu is set to On Demand.
3.
Set the corresponding Deactivation Timeout field with the delay, in seconds, to wait before
deactivating the bypass after an on hook if the bypass is activated on demand.
The delay is restarted after each on hook. The bypass is not deactivated if the delay expires while
the FXS is off hook.
This field is only used when the corresponding Activation drop-down menu is set to On Demand.
Note: For the Mediatrix 3308, Mediatrix 3316, Mediatrix 3731 and Mediatrix 3741 models, the bypass is
automatically disabled after the timeout, whether or not the bypass is in use. Furthermore, if the conditions
that activate the bypass are no longer present, the bypass is disabled.
4.
Click Submit if you do not need to set other parameters.
FXS Emergency Call Override
Note: This feature is available for FXS units only.
This FXS Emergency Call Override feature allows you to override a set of services that are activated during
an emergency call.
Two variables are available:


FxsEmergencyCallOverride : to override or not the services.
FxsEmergencyRingTimeout: to set the period before the phone starts to ring in the event where
the originator of an emergency call hangs-up before the emergency call center disconnects the
call.
The configuration of the Emergency Call Override is only available in the MIB parameters of the Mediatrix unit.
You can configure the parameters as follows:



by using a MIB browser
by using the CLI
by creating a configuration script containing the configuration variables
 To set the Emergency Call Override:
136
1.
In the potsMIB, set the FxsEmergencyCallOverride variable to the proper value, or
2.
In the CLI or a configuration script use:
Dgw v2.0 Application
FXS Distinctive Ring
Software Configuration Guide
Pots.FxsEmergencyCallOverride="Value"
where Value may be one of the following:
Table 95: Fxs Emergency Call Override Parameters
Value
Parameter
Description
100
NoOverride
The set of services for emergency calls remains the same as
configured. This is the default value.
200
NoServices
Ignores any service requiring a flash-hook. Call waiting and all other
related services are deactivated.
300
NoDisconnect
Ignores any service requiring a flash-hook. Call waiting and all other
related services are deactivated AND automatically re-establishes a
call that was disconnected by the originator.
 To set the Emergency Ring Timeout Override:
1.
Make sure the FxsEmergencyCallOverride variable is set to NoDisconnect.
2.
In the PotsMIB, set the FxsEmergencyRingTimeout variable to the proper value, or
3.
In the CLI or a configuration script use:
Pots.FxsEmergencyRingTimeout="Value"
where Value is in milliseconds. The default value is 2000 ms.
FXS Distinctive Ring
This FxsDistinctiveRingId parameter allows you to create a custom distinctive ring. Configuring the custom
distinctive ring allows the administrator to modify the ring pattern. When a pots.fxsDistinctiveRing.RindId is
defined, the corresponding ring pattern is used.
To use the distinctive ringing with the unit, the received SIP INVITE message must contain the Alert-Info
header field with the proper Call Property value.
Example
Alert-Info: <http://127.0.0.1/Bellcore-dr2>
Two variables are used to configure a distinctive ring:
•
RingId: Identifies the distinctive ring. When the incoming call property ‘distinctive-ring’
matches the defined RingId, the corresponding ring pattern is used. Otherwise, the
country ring pattern is used.
•
Pattern: Describes a tone pattern.
The format of the pattern is as follows:
ring-pattern = [ states-section ]
states-section = on-state-description "," off-state-description [ "," on-state-description
"," off-state-description [ "," on-state-description "," off-state-description ] ]
on-state-description = time
off-state-description = time
time = 2*5DIGIT
Table 96: Tag description
Tag
ring-pattern
Dgw v2.0 Application
Description
String describing the pattern to use for the ring. An empty
string means no ring.
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Chapter 22 - POTS Configuration
FXS Distinctive Ring
Table 96: Tag description
Tag
Description
states-section
Description of the state of the ring. Up to 3 pairs of states
can be defined. They must be at least one state described if
the ring-pattern is not empty.
on-state-description
Description of a state playing a ring.
off-state-description
Description of a state not playing a ring.
time
The number of time in ms to perform the action of the state.
Range is from 0 to 32767 ms.
Examples:
•
No ring:""
•
Bellcore-dr2: "800,400,800,4000"
•
Bellcore-dr4: "300,200,1000,200,300,4000"
Table 97: Default mapping between call property and ring cadence
Call Property Value
Ring cadence in milliseconds (bold are on,not bold are
off.
//127.0.0.1/Bellcore-dr2
800, 400, 800, 400
//127.0.0.1/Bellcore-dr3
400, 200, 400, 200, 800, 4000
//127.0.0.1/Bellcore-dr3
300, 200, 1000, 200, 300, 4000
All other value or call properties not present
Country’s normal ring.
The parameters can be set:



by using a MIB browser
by using the CLI
creating a configuration script containing the configuration variables.
 To customise a distinctive ring:
1.
2.
In the potsMIB set:
•
Pots.FxsDistinctiveRingId variable in the FxsDistinctiveRing table
•
Pots.FxsDistinctivePattern variable in the FxsDistinctiveRing table.
•
or
Use the CLI or a configuration script:
•

138
Pots.FxsDistinctivering[index=value].RingId="Value"
•
Pots.FxsDistinctivering[index=value].Pattern="Value"
Index value can vary from 1 to 4.
Dgw v2.0 Application
FXO Configuration
Software Configuration Guide
FXO Configuration
The FXO Config page allows you to configure gateways-specific parameters.
Note: This web page is available only on the following models:
• Mediatrix 3208 / 3216
• Mediatrix 3308 / 3316
• Mediatrix 3716
• Mediatrix 3731
• Mediatrix 3732
• Mediatrix 3741
• Mediatrix 3742
• Mediatrix C730 / C733 / C731
The FXO port’s use depends on the FXS bypass state (as defined in the FXS Bypass section of the FXS
Configuration page – “FXS Bypass” on page 135):


The FXO port becomes disabled as IP-FXO gateway when the FXS bypass is active. In this
case:
•
calls from the IP side to the FXO port are discarded
•
calls coming from the SCN side on the FXO port are routed to FXS port #1.
The FXO port becomes enabled and available for calls when the FXS bypass is disabled.
If an event that activates the FXS bypass occurs and the FXO port is in use, the bypass activation waits until
the FXO port is no longer in use.
FXO Dialing Configuration
The FXO Dialing Configuration section allows you to set dialing parameters.
 To set FXO dialing parameters:
1.
In the web interface, click the POTS link, then the FXO Config sub-link.
Figure 55: POTS FXO Config Web Page
2
3
4
2.
Set the Pre Dial Delay field with the delay, in milliseconds, between the time the line is successfully
seized, or dial tone detected, and the moment the destination phone number is dialed.
Some country specifications include a mandatory pre-dial delay. In that case, the highest value
between that and the value set in this field is used.
A value of ‘0’ indicates to use a value that is specific to the country specification as set in the
TelIf.CountrySelection parameter.
3.
Dgw v2.0 Application
Set the Dial Tone Detection Mode drop-down menu with the proper behaviour.
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Chapter 22 - POTS Configuration
FXO Configuration
When dial tone detection is enabled, the unit waits for a dial tone on the FXO line before initiating
the dialling sequence. If no dial tone is detected, the line is considered as busy with an incoming
call. This mechanism helps avoid collisions between incoming and outgoing calls.
Table 98: Dial Tone Detection Modes
Parameter
Disable
Description
Dial tone detection is disabled.
Country Tone The unit tries to detect the tone specified for this purpose in the current country's
tone specification. Some country specifications omit this information. In that case,
the unit behaves as if the parameter is set to Disable.
The following table lists the default dial tone detection frequency and cadence for each supported
country.
Table 99: Default Dial Tone Detection
Cadence
Country
Frequency
ON (s)
OFF (s)
Austria
450 Hz
0.5
0.0
Czech Republic
425 Hz
0.3
0.0
France
440 Hz
0.5
0.0
Germany1/2
425 Hz
0.15
0.0
Italy
425 Hz
0.15
0.0
NorthAmerica1
440 Hz
0.5
0.0
Spain
425 Hz
0.15
0.0
Switzerland
425 Hz
0.5
0.0
4.
Set the Dial Tone Detection Timeout field with the value, in milliseconds, indicating how long the
unit waits for a dial tone before considering the line is busy with an incoming FXO call.
5.
Click Submit if you do not need to set other parameters.
FXO Answering Configuration
The FXO Answering Configuration section allows you to define how the FXO line must behave when
answering calls.
 To set FXO answering parameters:
1.
In the FXO Answering Configuration section, set the corresponding Wait Before Answering Delay
column’s FXO interface field with the waiting period, in milliseconds, before answering an incoming
FXO call.
If this delay expires before the caller ID signal is decoded, the call proceeds without caller ID
information.
If a minimal waiting period is required for the selected country, the highest of both values is used.
Figure 56: FXO Answering Configuration Section
1
2.
140
2
3
Set the corresponding Answering On Caller ID Detection column’s drop-down menu with the proper
behaviour.
Dgw v2.0 Application
FXO Configuration
Software Configuration Guide
This parameter enables answering upon caller ID detection instead of the waiting delay configured
in Step 1.
When enabled, an incoming FXO call is answered on the first occurrence of either:
3.
•
The reception of the caller ID signal.
•
The expiration of the delay configured by the Wait Before Answering Delay field and
the country wait before answering delay.
Set the corresponding Wait For Callee To Answer column’s drop-down menu with the proper
behaviour.
When the endpoint is set up for automatic call (see “Automatic Call” on page 387), enabling this
variable makes the endpoint wait until the called the party connection is established before
answering the incoming call.
4.
Dgw v2.0 Application
Click Submit if you do not need to set other parameters.
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Chapter 22 - POTS Configuration
FXO Configuration
FXO Incoming Call Configuration
This section allows you to define how each line behaves when there is an incoming call.
 To set FXO incoming call parameters:
1.
In the FXO Incoming Call Behavior section, set the corresponding Not Allowed Behavior column’s
drop-down menu with the proper behaviour.
Under certain circumstances (locked port, configuration, etc.), incoming FXO calls are not allowed.
When that is the case, the FXO endpoint behaves in one of the manners below.
Table 100: Not Allowed Behavior Parameters
Parameter
Description
Do Not Answer
The incoming call is left unanswered.
Play Congestion Tone
The incoming call is answered, a congestion tone is played for 10
seconds, and then the call is terminated.
Figure 57: FXO Incoming Call Behavior Section
1
2.
Click Submit if you do not need to set other parameters.
FXO Custom Basic Parameters
Overrides the default country basic parameters for this FXO line.
The parameters can be set:



by using a MIB browser
by using the CLI
by creating a configuration script containing the configuration variables.
 To override the FXO Custom Basic Parameters:
1.
In the potsMIB set:
•
OverrideDefautCountryParameters parameter in the FxoCustomBasicParameters table
•
2.
Impedance parameter in the FxoCustomBasicParameters table
Use the CLI or a configuration script:
•
Pots.FxoCustomBasicParameters[index=value].OverrideDefautCountryParameters="Value"
Where value can be:
enable: The FXO line uses the custom values as defined in the current row. To reset the parameter values to the default country values,
use the Reset row command.
disable: The FXO line uses the default country values. The values set in the current row are not applied.
•
Pots.FxoCustomBasicParameters[index=value].Impedance="Value"
Where value equals:
Table 101: Impedance Values
Value
142
Meaning
Description
100
I600
Impedance of 600 ohms.
(Used in North America)
200
I600LongLoop
ANSI/EIA/TIA 464 compromise
impedance for trunks.
Dgw v2.0 Application
FXO Configuration
Software Configuration Guide
Table 101: Impedance Values
Value
Meaning
Description
300
I900
Impedance of 900 ohms.
400
Australia
Impedance used in Australia
500
Austria
Impedance used in Austria
600
Belgium
Impedance used in Belgium
700
Brazil
Impedance used in Brazil
800
China
Impedance used in China
900
Czech Republic
Impedance used in Czech
Republic
1000
Denmark
Impedance used in Denmark
1100
Finland
Impedance used in Finland
1200
France
Impedance used in France
1300
Germany
Impedance used in Germany
1400
Greece
Impedance used in Greece
1500
Italy
Impedance used in Italy
1600
Japan
Impedance used in Japan
1700
Netherlands
Impedance used in
Netherlands
1800
New Zealand
Impedance used in New
Zealand
1900
Norway
Impedance used in Norway
2000
Russia
Impedance used in Russia
2100
Slovakia
Impedance used in Slovakia
2200
Spain
Impedance used in Spain
2300
Sweden
Impedance used in Sweden
2400
UK
Impedance used in UK
FXO Line Verification
The line state verification mechanism allows the detection of defective or down lines based on the absence of
current when closing the loop. You can view the line state in the FXO Line State table (“FXO Line Status” on
page 128).
 To set FXO line verification parameters:
1.
In the FXO Line Verification section, set the Link State Verification drop-down menu with the proper
behaviour.
Figure 58: FXO Answering Configuration Section
2
3
Dgw v2.0 Application
143
Chapter 22 - POTS Configuration
FXO Configuration
2.
Set the Link State Verification Timeout field with the value, in milliseconds, indicating how long the
unit waits to successfully take the line before considering the line is defective or down.
3.
Click Submit if you do not need to set other parameters.
FXO Force End of Call
FXO Force end of call, also known as Far End Disconnect, refers to methods for detecting that a remote party
has hung up. If the Far End Disconnect signal is not sent to or properly detected by the Mediatrix unit, the
connection will not be released by the unit, thus freezing the FXO line in the off hook state.
The FXO Force End of Call section allows you to define various methods to detect a far end disconnect.
 To set FXO force end of call parameters:
1.
In the FXO Force End of Call section, set the Force End Of Call On Call Failure drop-down menu
with the proper behaviour.
This parameter enables or disables forced-end-of-call on call failure.
The forced end of call on call failure occurs when a SCN caller tries to reach someone on the IP
network and the SCN caller hangs up before reaching the IP callee.
If the connection is not established, the Mediatrix unit cannot detect that the caller has hung up. You
can configure the Mediatrix unit to disconnect an unsuccessful communication after a specific
number of seconds. This number of seconds is defined in a timeout that starts to count when the
SCN caller contacts the Mediatrix unit. If the connection is not successful, the line is closed when
the call time reaches the defined timeout value.
This feature forcefully terminates a call that stayed in an error state for some time. When the line
falls in an error state where a SIT, a ROH, a BUSY or any error tone is played outbound to the FXO
line, the unit waits for the timeout specified in the Call Failure Timeout field and then hangs up.
Figure 59: FXO Force End of Call Section
1
3
5
2
4
6
7
8
2.
Set the Call Failure Timeout field with the waiting period, in seconds, before terminating a call in an
error state.
3.
Set the Forced End Of Call On Silence Detection Mode drop-down menu with the proper behaviour.
This parameter enables or disables forced-end-of-call on silence detection.
The silence detection feature applies when two parties are in communication and one of them
hangs up. It allows the Mediatrix unit to close a line when no voice activity or silence is detected for
a specified amount of time.
When silence is detected on the IP and/or SCN side for an amount of time specified in a timeout,
the call is terminated. This feature is useful to free resources in the event of an IP network failure
that prevents the end of call to be detected or when the SCN end of call tone was not detected.
Note: The silence detection feature could inadvertently disconnect a communication when one party puts
the other on hold. Using the hold tone can prevent detection of silence when the call is put on hold by the IP
peer. See “Tone Override” on page 412 for more details.
The current implementation of silence detection relies on the power of the media signal. A silence
is detected if the power level of the media signal is lower than -60 dBm.
144
Dgw v2.0 Application
FXO Configuration
Software Configuration Guide
This feature forcefully terminates a call that stayed silent for some time. When silence is detected
on the inbound and/or outbound media for an amount of time specified in the Silence Detection
Timeout field, the call is terminated.
Table 102: Forced End of Call on Silence Detection Mode Parameters
Parameter
Disable
Description
Forced-end-of-call on silence detection is disabled.
Inbound And Outbound Silent The call is terminated if both inbound and outbound media are
silent at the same time.
This feature is useful to free resources in the event of a network failure preventing the end-of-call to
be detected or when the FXO end-of-call tone was not detected.
4.
Set the Silence Detection Timeout field with the maximum amount of time, in seconds, that a call
can remain silent, before it is terminated and the line is released.
5.
Set the Forced End Of Call On Tone Detection Mode drop-down menu with the proper behaviour.
This parameter enables or disables forced-end-of-call upon tone detection. It terminates a call upon
detection of an end-of-call tone on the inbound from the FXO line.
Table 103:
Parameter
Description
Disable
Force-end-of-call upon tone detection is disabled.
Country
Tone
The unit tries to detect the tone specified for this purpose in the current country's
tone specification. Some country specifications omit this information. In that case,
the unit behaves as if the parameter is Disable.
Custom Tone Terminates a call upon detection of a custom tone. See Steps 6-8.
The Mediatrix unit can monitor special tones that indicate the remote SCN user has hung up. You
can use the default value for a selected country or customize a tone. Table 104 lists the default
frequency and cadence detected by supported country.
Table 104: Default Frequency and Cadence Supported
Cadence
Country
Frequency
ON1 (s)
OFF1 (s)
ON2 (s)
OFF2 (s) Repetition
Austria
450 Hz
8.0
0.0
0.0
0.0
1
Czech Republic
425 Hz
8.0
0.0
0.0
0.0
1
France
440 Hz
8.0
0.0
0.0
0.0
1
Germany
425 Hz
8.0
0.0
0.0
0.0
1
Italy
425 Hz
0.2
0.2
0.6
1.0
4
North America
440 Hz
8.0
0.0
0.0
0.0
1
Spain
425 Hz
8.0
0.0
0.0
0.0
1
Switzerland
425 Hz
8.0
0.0
0.0
0.0
1
When customizing a tone, Media5 suggests to ask your Central Office about the tone it generates
to indicate a call has been disconnected. You will thus be able to customize your tone according to
this information.
Custom Tone Settings
Dgw v2.0 Application
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Chapter 22 - POTS Configuration
6.
FXO Configuration
If you have selected Custom Tone in Step 5, set the Tone Detection Custom Frequency field with
the Frequency, in Hertz (Hz), to detect in the custom cadence.
You can set any value between 350 Hz and 620 Hz.
A customized tone detection can only detect a single frequency. To detect tones made of multiple
frequencies, create the cadence for only one of the frequencies found in the tone.
7.
Set the Tone Detection Custom Cadence field with the cadence to detect.
A cadence is a series of frequencies that are played for a specified time, making up a tone. The
format for a cadence is:
on1,off1,on2,off2,on3,off3
In this string, “on” and “off” are numerical values representing the time, in milliseconds, that the
frequency can and cannot be detected, respectively. For instance, “2000, 1000, 2000, 0” is a
cadence in which the frequency plays for 2 seconds, stops for 1 second, and plays for 2 more
seconds. This example is also equivalent to setting the string “2000, 1000, 2000”.
You can specify up to three “on,off” pairs. If you specify less than those six values, “0” values will
be added as necessary. Specifying more than six will only use the six first values.
A cadence starting with a value of zero (0) is invalid. The first zero (0) found in the string signals the
end of the cadence (i.e. “200, 0, 300” is the same as “200”).
•
To detect a continuous tone, use a single “on” value, e.g., “200, 0” or “200” (the off time
is 0 ms.). A continuous tone of 200 ms is used if the field is empty.
•
To detect a tone in which two or more frequencies are used, for instance with a
cadence of “200 ms, 500 ms, 300 ms, 400 ms”, in which the respective frequencies
would be “400 Hz, 300 Hz, 400 Hz, 500 Hz”, detect a frequency that comes twice or
more in the tone. In the above example, detect the 400 Hz frequency by using the
cadence “200 ms, 400 ms, 300 ms, 500 ms”. In this example, the 400 ms and 500 ms
off times represent the times that the 400 Hz frequency cannot be heard, even though
another frequency may be playing.
The “on” and “off” values can be from 0 to 32767 ms.
146
8.
Set the Detection Custom Repetition field with the number of times the custom cadence must be
detected to consider the custom end-of-call tone has been detected.
9.
Click Submit if you do not need to set other parameters.
Dgw v2.0 Application
ISDN Parameters
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Introduction
C
Software Configuration Guide
H A P T E R
23
ISDN Configuration
This chapter describes how to configure the Integrated Services Digital Network (ISDN) Basic Rate Interfaces
(BRI) and/or Primary Rate Interfaces (PRI) parameters of the Mediatrix unit.
Note: This web page is not available on the Mediatrix LP/4100/C7 Series models.
Introduction
ISDN is a set of digital transmission protocols defined by a few international standards body for
telecommunications, such as the ITU-T. One or another of these protocols are accepted as standards by
virtually every telecommunications carrier all over the world.
ISDN replaces the traditional telephone system so that one or two pairs of telephone wires can carry voice and
data simultaneously. It is a fully digital network where all devices and applications present themselves in a
digital form.
ISDN is a User-Network Interface (UNI) signalling protocol with a user and a network side. The user side is
implemented in ISDN terminals (phones, terminal adapters, etc.) while the network side is implemented in the
exchange switches of the network operator. Both sides have different signaling states and messages. The
Mediatrix unit ISDN interfaces can be configured to work as user (TE) or network (NT) interfaces.
Depending on your product, you can configure two types of ISDN interfaces:

The ISDN Basic Rate Interface (BRI) – Mediatrix 3404, 3408, 3734, 3741, 3742, and 4400
Series models.

The ISDN Primary Rate Interface (PRI) – Mediatrix 3531, 3532, 3621, 3631, 3632, 3731, 3732,
and 3734 models.
ISDN Reference Points
ISDN specifies a number of reference points that define logical interfaces between the various equipment
types on an ISDN access line. The Mediatrix unit supports the following ISDN reference points:

S: The reference point between user terminals and the NT2. This is used in point-to-multipoint
BRI connections.

T: The reference point between NT1 (Modem) and NT2 (PBX) devices. This is used in pointto-point PRI/BRI connections.
All other ISDN reference points are not supported.
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Introduction
Figure 60: ISDN Reference Points
BRI point-to-point
S
T
NT2
TE
NT1
BRI point-to-multipoint (S-bus)
ISDN
S/T
TE
TE
NT1
PRI
S
T
NT2
TE
NT1
Inband Tones Generation
In an ISDN network, most of the call setup tones are played locally by the TE equipment (i.e., telephone
handset), although some require that the tones be played inband by the NT.
When interworking with other networks occurs, the need for the tones to be played inband is more likely to
arise.
The Mediatrix unit may enable inband tones to be played locally, on a per-interface basis. This option is
present when the unit is acting as both the NT and the TE UNI-side. However, in TE mode, only the ringback
tone is played.
The Call Setup tones (dial tone, ringback tone, etc.) are played in the direction where the call has been
initiated. The call disconnection tones are played in both directions, but of course will not arrive to the peer
who disconnected the call.
When an inband tone is played, a Progress Indicator information element (IE) #8 “Inband information or
appropriate pattern available” is added to the ISDN message corresponding to the call state change, and in a
PROGRESS ISDN message if no state change is occurring.
When an interface is acting as the TE, as soon as the NT advertises that it plays inband tones through a
Progress Indicator IE #8 or #1, the local inband tones generation is disabled for the rest of the call.
Whenever a tone is played inband locally or the ISDN peer advertises that inband informations are available,
the Mediatrix unit is notified. The IP media path can then be opened earlier in the call, and can be closed with
some delay after the call disconnection initiation.
The following table summarizes the inband tones generation behaviour for the NT mode.
Table 105: Inband Tones Generation Behaviour - NT
Inband Tones Generation
Enabled
Signal IE Handling Enabled
Inband Tone Played
No
No
No
No
Yes
Yes
Yes
Don’t care
No
The following table summarizes the inband tones generation behaviour for the TE mode.
Table 106: Inband Tones Generation Behaviour - TE
Signal IE
Handling Enabled
No
150
Signal IE
Received
Don’t care
Inband Tones
Generation
Enabled
No
NT Peer
Advertised
Inband Tones
Don’t care
Inband Tone
Played
No
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Table 106: Inband Tones Generation Behaviour - TE (Continued)
Signal IE
Handling Enabled
Inband Tones
Generation
Enabled
Signal IE
Received
NT Peer
Advertised
Inband Tones
Inband Tone
Played
Yes
Yes
No
Don’t care
Yes
Yes
No
Don’t care
Don’t care
No
No
Don’t care
Yes
Yes
No
Note that when the PRI/BRI interface Signalling Protocol drop-down is set to QSIG, the Signal IE does not
exist so it has no effect on the inband tones generation. In QSIG, inband tones are played when the inband
tones generation is activated on the incoming side of the call. See “PRI Configuration” on page 155 or “BRI
Configuration” on page 167 for more details.
Signal Handling
The Signal IE is used by the NT ISDN side to tell its TE peers that they must generate an inband tone locally.
Thus, the Signal IEs are sent by the NT only.
When the Signal IE handling is enabled on a given ISDN interface acting as a TE, inband tones are played
towards the IP side when a Signal IE is received. On a NT interface, a Signal IE is inserted in the ISDN
messages sent to the TE when appropriate.
Note that when the Mediatrix 3500 Series signaling protocol is set to “NI-2” (National ISDN-2) on that interface,
the Signal IE handling is forced to be enabled for a NT.
Setting PRI Hardware Parameters
You must set hardware-related parameters. You can do so in the System / Hardware page. The Hardware
page differs depending on the product and model you have.
 To configure the Mediatrix unit hardware:
1.
In the web interface, click the System link, then the Hardware sub-link.
Figure 61: System – Hardware Web Page
2
2.
3
4
In the PRI Cards Configuration section, select the reference of the clock source in the Clock
Reference drop-down menu.
If you want to configure the clock reference of a specific interface, you must set the Endpoint Type
drop-down menu to NT. See “PRI Configuration” on page 155 for more details.
Table 107: Clock Reference
Reference
None
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Description
The internal clock does not synchronize with any other source.
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Table 107: Clock Reference (Continued)
Reference
Other
Card
Description
The internal clock synchronizes with the other PRI interface of the Mediatrix unit.
This interface must be configured in TE mode (Endpoint drop-down menu of the
Interface Configuration section) to provide the clock reference to the other
interfaces.
Note: This choice is not available on the Mediatrix 3531, 3621 and 3631 models.
3.
Select whether the line uses T1 or E1 in the Line Type drop-down menu.
You must restart the unit if you change this setting.
4.
Select the ISDN signaling in the Signaling drop down menu.
When changing from R2 to ISDN or ISDN to R2, you must change your routes accordingly. For
instance, if you are in R2 with a route r2-Slot2/E1T1, then change to ISDN, you must change the
route to isdn-Slot2/E1T1.
This parameter is available only for the Mediatrix 3621/3631/3632 models. Other models support
only the ISDN protocol.
5.
Click Submit if you do not need to set other parameters.
ISDN Auto-Configuration
The ISDN Auto-configuration feature allows you to detect and to configure all ISDN interfaces so that the ISDN
link goes up and becomes usable with a minimal user interaction. When launching an auto-configuration
process, it stops automatically when all interfaces have been tested. For each interface, the autoconfiguration process is considered successful when the link becomes up or a failure when all combinations
have been tried without having a link up.
Caution: Launching the auto-configuration may terminate abruptly all ongoing ISDN calls.
Note: Auto-configuration on all ISDN interfaces may take some time to complete. Some of the current ISDN
settings might be replaced by new values.
Please note that some parameters cannot be auto configured. For instance, the clock mode is configured
according to the endpoint type, master for NT and slave for TE.
 To launch the auto-configuration process:
1.
In the web interface, click the ISDN link, then the Status sub-link.
Figure 62: ISDN – Status Configuration Section
2.
Click the Start Sensing button.
The process starts.
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Preset
Software Configuration Guide
Preset
The ISDN Preset Configuration section allows you to load a set of preset configuration for your ISDN
connections. These preset files are located in the file system's persistent memory. They differ depending on
the Mediatrix unit you are using. Depending on your unit's profile, it may be possible that no preset files are
available.
Using preset files is especially useful for units that do not use the default values provided by Media5 (for
instance, T1 instead of E1 for Mediatrix 3000 units). Please note that only script files work. Any other type of
file present in the file system cannot be run here.
You can also export your current ISDN configuration in a preset. Please note that these user-defined presets
are not kept in the event of a partial or factory reset.
To see the content of the unit’s file system persistent memory, go to File Manager (“Chapter 53 - File Manager”
on page 543). All installed configuration scripts/images are listed.
 To load and execute a preset file:
1.
In the ISDN Status tab, ISDN Preset Configuration section, select one of the available preset files
in the Local Preset drop-down menu.
Figure 63: ISDN – Status Configuration Section
1
2.
Click Apply.
The configuration is applied.
 To export the current ISDN configuration as a preset:
1.
In the ISDN Preset Configuration section, type a name for the preset in the Preset Name field.
Figure 64: ISDN – Status Configuration Section
1
2.
Click Save.
The current ISDN configuration is exported. Please note that these user-defined presets are not
kept in the event of a partial or factory reset.
When the clock device is not synchronized, the description value of the file is "Automatically
Generated". When synchronized, the description is "Automatically Genereted on Date/Time". See
the File Manager (“Chapter 53 - File Manager” on page 543) for more details on how to see and
manage the files in the unit’s file system.
Partial Reset
When a partial reset is triggered, the user-defined presets are deleted.
PRI ISDN Statistics
The Mediatrix unit collects meaningful statistics for each PRI digital card that can be read via the web interface.
These statistics are also available via SNMP and CLI.
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PRI ISDN Statistics
The Mediatrix unit collects statistics for each of its two cards, if available. Slot 2 and Slot 3 indicate the physical
location of the cards in the unit, Slot 2 being on the left when looking at the rear of the unit. You can click the
Reset Stats button at any time to reset all statistics for the specified interface.
Figure 65: ISDN – Statistics Web Page
The following table describes the statistics available.
Table 108: ISDN Statistics Displayed
Statistic
TxFrames
Description
Number of HDLC frames transmitted.
Note: The term frames does not refer to the structure defined in I.431.
RxFrames
Number of HDLC frames received.
Note: The term frames does not refer to the structure defined in I.431.
TxOctets
Number of octets transmitted. This value is obtained by cumulating the octets
transmitted in the HDLC frames.
Note: The term frames does not refer to the structure defined in I.431.
RxOctets
Number of octets received. This value is obtained by cumulating the octets received
in the HDLC frames.
Note: The term frames does not refer to the structure defined in I.431.
FcsErrors
Frame check sequence (FCS) errors indicate that frames of data are being
corrupted during transmission. FCS error count is the number of frames that were
received with a bad checksum (CRC value) in the HDLC frame. These frames are
dropped and not propagated in the upper layers.
This value is available on E1 and T1.
FramesDropped
Number of frames dropped. This value is obtained by cumulating the number of
frames dropped when transferring the data from the framer chip to the device
internal buffer.
This value is available on E1 and T1.
OctetsDropped
Number of octets dropped. This value is obtained by cumulating the number of
octets dropped when transferring the data from the framer chip to the device internal
buffer.
This value is available on E1 and T1.
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Table 108: ISDN Statistics Displayed (Continued)
Statistic
NegFrameSlipsTx
Description
A frame is skipped when the frequency of the transmit clock is greater than the
frequency of the transmit system interface working clock based on 2.048 MHz (on
E1) or 1.544 MHz (on T1).
This value is available on E1 and T1.
NegFrameSlipsRx
A frame is skipped when the frequency of the received route clock is greater than
the frequency of the receive system interface working clock based on 2.048 MHz (on
E1) or 1.544 MHz (on T1).
This value is available on E1 and T1.
PosFrameSlipsTx
A frame is repeated when the frequency of the transmit clock is less than the
frequency of the transmit system interface working clock based on 2.048 MHz (on
E1) or 1.544 MHz (on T1).
This value is available on E1 and T1.
PosFrameSlipsRx
A frame is repeated when the frequency of the receive route clock is less than the
frequency of the receive system interface working clock based on 2.048 MHz (on
E1) or 1.544 MHz (on T1).
This value is available on E1 and T1.
FramingErrors
The framing error count indicates that a FAS (Frame Alignment Signal) word has
been received with an error.
The FAS-bits are present in every even frame of timeslot 0 on E1.
The FAS-bits are present in ESF format on T1.
This value is available on E1 and T1.
CodeViolations
The code violations count indicates that an encoding error on the PCM line has been
detected.
This value is available on E1 and T1.
CRCErrors
The CRC errors count is incremented when a multiframe has been received with a
CRC error.
The CRC error count is available in CRC multiframe mode only on E1.
The CRC error count is in ESF format on T1.
EBitErrors
The E-Bit error count gives information about the outgoing transmit PCM line if the
E-bits are used by the remote end for submultiframe error indication. Incrementing is
only possible in the multiframe synchronous state.
Due to signaling requirements, the E-bits of frame 13 and frame 15 of the CRC
multiframe can be used to indicate an error in a received submultiframes:
Submultiframe I status E-bit located in frame 13
Submultiframe II status E-bit located in frame 15
no CRC error : E = 1
CRC error : E = 0
This value is only available in E1.
BlockErrors
The Block Error count is incremented once per multiframe if a multirame has been
received with a CRC error or a bad frame alignment has been detected.
This value is only available for ESF format on T1 only.
PRI Configuration
This section applies to the following models:
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Chapter 23 - ISDN Configuration








PRI Configuration
Mediatrix 3531
Mediatrix 3532
Mediatrix 3621
Mediatrix 3631
Mediatrix 3632
Mediatrix 3731
Mediatrix 3732
Mediatrix 3734
The Primary Rate Interface (PRI) port supports 30 x 64 kbit/s B-channels, 1 x 64 kbit/s D-channel and 1 x
synchronization timeslot on a standard E1 (G.704) physical layer. In its T1 version, a PRI interface supports
23 x B-channels and 1 x D-channel, all at a 64 kbit/s rate. E1 is mostly deployed in Europe, while T1 is more
present in North America.
Caution: You can configure ISDN ports while they are active. However they are internally disabled to
modify the configuration and then re-enabled. All active calls on the port are dropped during this process.
Configuration changes should only be performed during planned down times. Most of the ISDN parameters
change require a restart of the ISDN service to be applied.
Caution: The Mediatrix unit PRI ports can be used as a T reference point, but not as U reference points
(2-wire). Never connect a U SCN line or a U TE into the Mediatrix unit PRI ports.
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 To configure the PRI parameters:
1.
In the web interface, click the ISDN link, then the Primary Rate Interface sub-link.
Figure 66: ISDN – Interface Configuration Section
2
4
6
8
10
12
5
7
9
11
13
14
15
16
17
18
20
19
21
22
24
23
25
26
28
27
29
30
32
31
33
34
36
38
35
37
2
2.
Select to which interface you want to apply the changes in the Select Interface drop-down menu at
the top of the window.
You can copy the configuration of the selected interface to one or more interfaces of the Mediatrix
unit in the Apply to the Following Interfaces section at the bottom of the page. You can select
specific interfaces by checking them, as well as use the Check All or Uncheck All buttons.
The Mediatrix 3532 and 3632 models have two interfaces.
3.
If applicable, use the Line Type Configure link to set the line type, as described in “Setting PRI
Hardware Parameters” on page 151.
4.
Select the endpoint type in the Endpoint Type drop-down menu.
Table 109: Endpoint Type
Type
TE
Dgw v2.0 Application
Description
Terminal Equipment.The endpoint emulates the subscriber (terminal) side of the digital
connection. You can connect the SCN to the endpoint.
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Table 109: Endpoint Type (Continued)
Type
NT
Description
Network Termination. The endpoint emulates the central office (network) side of the
digital connection. You can connect a PBX to the endpoint.
The setting used for the Mediatrix unit must be opposite to the setting used in the PBX. For instance,
if the PBX is set to TE, then the Mediatrix unit must be set to NT.
Note: If you want to use a specific interface as the reference clock, you must set it to TE.
When the PRI interface Signalling Protocol drop-down is set to QSIG (see Step 9), the endpoint type
is only used in the second layer (LAPD) since it is a concept that does not exist in QSIG.
5.
Select the clock mode of the interface in the Clock Mode drop-down menu.
The interface can either generate the clocking for the line or accept the clock from the line.
Table 110: Clock Mode
Mode
auto
Description
The setting is derived from the endpoint type.
•
NT: clock master
•
TE: clock slave
Master The interface generates the clock.
Slave
The interface accepts the clock from the line.
The clock mode is used to give the user the possibility to set an endpoint in TE mode and still
generate the clock by specifying the clock mode to master. The clock source can then be selected
from the Clock Reference drop-down menu (see “Setting PRI Hardware Parameters” on page 151
for more details). The clock mode could be used, for instance, to synchronize several units in NT
mode via a T1 or E1 line.
6.
Select the port pinout in the Port Pinout drop-down menu.
Table 111: Port Pinout
Mode
7.
Description
Auto
The pinout is set according to the Endpoint Type parameter setting (Step 4).
Te
Forces the pinout to TE regardless of the Endpoint Type value.
Nt
Forces the pinout to NT regardless of the Endpoint Type value.
Set the Monitor Link State drop-down menu with the physical link state of the ISDN interface.
Table 112: Interface Link State
Parameter
Description
Enable
The ISDN endpoint's operational state is affected by its interface physical link state.
When the link state of the ISDN interface is down, the operational state of its
matching endpoint becomes “disable”.
Disable
The ISDN endpoint's operational state is not affected by its interface physical link
state.
Note that if the Monitor Link State parameter is enabled and the Ignore SIP OPTONS on no usable
endpoints parameter is also enabled in the SIP / Interop page, this will influence how the SIP options
are answered. See “SIP Interop” on page 279 for more details.
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8.
Select the transmission encoding of bits in the Line Coding drop-down menu.
Table 113: Transmission Encoding
Coding
Description
B8ZS
Bipolar with 8-Zeros Substitution (T1 lines).
HDB3
High-Density Bipolar with 3-zeros (E1 lines).
AMI
Alternate Mark Inversion (E1 and T1 lines).
Make sure that the transmission encoding matches with the remote system. For further information,
see ITU-T Recommendation G.703.
9.
Select the frame format in the Line Framing drop-down menu.
Line Framing is used to synchronize the channels on the frame relay circuit (when a frame starts
and finishes). Without it, the sending and receiving equipment would not be able to synchronize
their frames.
Table 114: Line Framing
Format
Description
SF
Super frame. Sometimes known as D4 (T1 lines).
ESF
Extended super frame (T1 lines).
CRC4
Cyclic redundancy check 4 (E1 lines).
NO-CRC4 No Cyclic redundancy check 4 (E1 lines).
For further information, see ITU-T Recommendation G.704.
10.
Select the protocol to use for the signalling channel in the Signalling Protocol drop-down menu.
This signalling must match the connected ISDN equipment or network.
Table 115: Signalling Protocols
Protocol
Description
DSS1
Digital Subscriber Signaling System No.1
DMS100
Digital Multiplex System 100
NI2
National ISDN No.2
5ESS
5 Electronic Switching System
QSIG
ECMA's protocol for Private Integrated Services
Networks
The 5ESS and DMS100 protocols support basic call only and the model is derived directly from the
DSS1 protocol.
If you select the NI2 protocol, see “InformationFollowing Operation” on page 166 for the behaviour
if this parameter is present.
11.
Select the value of the network location in the progress indicator messages that the unit sends in
the Network Location drop-down menu.
This defines the location code to be inserted in ISDN causes code information elements, i.e., the
type of network to which the system belongs. The following values are available:
Dgw v2.0 Application
•
User
•
Private
•
Public
•
Transit
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•
12.
International
Set the Preferred Encoding Scheme drop-down menu with the data encoding scheme in the bearer
capabilities (user information layer 1 protocol).
This encoding scheme is used when initiating a call on the ISDN side. The supported encoding
schemes are G.711 u-Law and G.711 a-Law.
G.711 u-Law may not be supported by DSS1 NT and TE endpoints. It is recommended to use G.711
a-Law as preferred encoding protocol.
13.
Set the Fallback Encoding Scheme drop-down menu with the fallback data encoding scheme in
case the preferred encoding scheme is not available.
The supported encoding schemes are G.711 u-Law and G.711 a-Law.
Note: The fallback encoding scheme is valid only when receiving a SETUP message. The user sending the
SETUP message does not indicate alternative bearer capability.
If the proposed encoding scheme in the bearer capability received in the SETUP message is
different than the preferred encoding scheme, then the fallback encoding scheme is used.
14.
Define the range of active bearer channels in the Channel Range field.
15.
Define the range to reserve channels for incoming calls in the Channels Reserved for Incoming
Calls field.
Bearer channels are by default usable for both incoming and outgoing calls. Use this range to
reserve channels for incoming calls.
Note:
• Channels outside of the range defined by the Channel Range field are ignored.
• Channels reserved in both the Channels Reserved for Incoming Calls and Channels Reserved for
Outgoing Calls fields are considered usable for both incoming and outgoing calls.
The string has the following syntax:
•
',': Separator between non-consecutive lists of channels or single channels.
•
'n': A single channel, where n is the channel number.
•
'm-n': List of channels where m is the start channel number and n is the end channel
number.
Note: The space character is ignored and duplication is not allowed. Channels must be specified in low to
high order.
Example: '1,12-15': The accepted channels are 1, 12, 13, 14 and 15.
16.
Define the range to reserve channels for outgoing calls in the Channels Reserved for Outgoing Calls
field.
Bearer channels are by default usable for both incoming and outgoing calls. Use this range to
reserve channels for outgoing calls.
Note:
• Channels outside of the range defined by the Channel Range field are ignored.
• Channels reserved in both the Channels Reserved for Incoming Calls and Channels Reserved for
Outgoing Calls fields are considered usable for both incoming and outgoing calls.
The string has the following syntax:
160
•
',': Separator between non-consecutive lists of channels or single channels.
•
'n': A single channel, where n is the channel number.
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Software Configuration Guide
•
'm-n': List of channels where m is the start channel number and n is the end channel
number.
Note: The space character is ignored and duplication is not allowed. Channels must be specified in low to
high order.
Example: '1,12-15': The accepted channels are 1, 12, 13, 14 and 15.
17.
Select the strategy for selecting bearer channels in the Channel Allocation Strategy drop-down
menu.
Table 116: Channel Allocation Strategy
Allocation
Description
Ascending
Starting from the lowest-numbered non-busy bearer channel and
going toward the highest-numbered non-busy bearer channel, the
Mediatrix unit selects the first bearer channel available.
Descending
Starting from the highest-numbered non-busy bearer channel and
going toward the lowest-numbered non-busy bearer channel, the
Mediatrix unit selects the first bearer channel available.
RoundRobinAscending
The Mediatrix unit starts from the bearer channel that follows the
bearer channel used for the last call. For instance, if channel #1 was
used in the last call, the unit starts with channel #2. Going toward the
highest-numbered non-busy bearer channel, the unit selects the first
channel available. If the highest channel is unavailable, the search
continues from the lowest-numbered non-busy bearer channel.
RoundRobinDescending The Mediatrix unit starts from the bearer channel that precedes the
bearer channel used for the last call. For instance, if channel #3 was
used in the last call, the unit starts with channel #2. Going toward the
lowest-numbered non-busy bearer channel, the unit selects the first
channel available. If the lowest channel is unavailable, the search
continues from the highest-numbered non-busy bearer channel.
18.
Define the maximum number of active calls on the interface in the Maximum Active Calls field.
This limits the total number of concurrent calls on the interface. Entering 0 indicates no maximum
number of active calls.
19.
20.
Select whether or not the signal information element is enabled in the Signal Information Element
drop-down menu.
•
When activated at the Network UNI-side (Endpoint Type drop-down menu set to NT),
the signal information element is sent to the User UNI-side.
•
When activated at the User UNI-side (Endpoint Type drop-down menu set to TE), the
tone indicated in the signal information element is played in the IP direction.
Select whether or not inband tone generation is enabled in the Inband Tone Generation drop-down
menu.
When activated at the User UNI-side (Endpoint Type drop-down menu is set to TE), only the
ringback tone is generated.
When the Signalling Protocol drop-down menu is set to QSIG (Step 9) and this variable is activated,
the incoming side of the call plays the tones inband.
21.
Select whether or not inband DTMF dialing is enabled in the Inband DTMF Dialing drop-down menu.
If you select Enable, the Mediatrix unit accepts inband DTMF digits when Overlap Dialing occurs.
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22.
PRI Configuration
Select whether or not overlap dialing is enabled in the Overlap Dialing drop-down menu.
Table 117: Overlap Dialing Parameters
Parameter
23.
Description
Enable
The Mediatrix unit transports the called-party number digit by digit, after the first
SETUP message, which contains no called party information at all.
Disable
The Mediatrix unit transports the full called party information in the first SETUP
message from the terminal. This means that the user must dial the number before
going off-hook.
Define the maximum length of the calling party name for calls from SIP to ISDN in the Calling Name
Max Length field.
Available values range from 0 to 82.
24.
Select whether or not exclusive B-Channel selection is enabled for calls from SIP to ISDN in the
Exclusive B-Channel Selection drop-down menu.
If you select Enable and initiate a call, only the requested B channel is accepted; if the requested
B channel is not available, the call is cleared.
25.
Select whether or not to enable the Sending Complete information element into SETUP messages
for calls from SIP to ISDN in the Sending Complete drop-down menu.
Some ISDN switches may require that the Sending Complete information element be included in
the outgoing SETUP message to indicate that the entire number is included and there are no further
destination digits to be sent.
26.
Select whether or not to enable sending the RESTART message upon a signalling channel “UP”
event in the Send Restart On Startup drop-down menu.
The RESTART message requests a restart for the interface specified.
27.
Set the link establishment strategy in the Link Establishment drop-down menu.
Table 118: Link Establishment Parameters
Parameter
Description
OnDemand When the data link is shut down, the unit establishes a new link only when
required.
Permanent
28.
When the data link is shut down, the unit immediately attempts to establish a new
link.
Set the STATUS causes that can be received without automatically clearing the call in the Accepted
Status Causes field.
The default action is to clear the call upon receiving a STATUS message. If a STATUS message is
received indicating a compatible call state and containing the supplied STATUS causes, the
clearing of the call is prevented.
The string has the following syntax:
•
',': Separator between non-consecutive lists of causes or single cause.
•
'n': A single cause, where n is the cause number.
•
'm-n': List of causes where m is the start cause number and n is the end cause number.
Note: The space character is ignored and cause duplication is not allowed.
Causes must be specified in low to high order.
Example: '1,124-127': The accepted causes are 1, 124, 125, 126 and 127.
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29.
Set the range of PROGRESS causes accepted by the unit in the Accepted Progress Causes field.
Causes excluded from this range trigger call disconnections.
The string has the following syntax:
•
',': Separator between non-consecutive lists of causes or single cause.
•
'n': A single cause, where n is the cause number.
•
'm-n': List of causes where m is the start cause number and n is the end cause number.
Note: You must consider the following:
• The space character is not allowed.
• Causes must be specified in low to high order.
• Cause duplication is not allowed.
Example: '1,124-127': The accepted causes are 1, 124, 125, 126 and 127.
30.
Select the strategy for sending ISDN Progress messages in the Send Isdn Progress drop-down
menu.
Table 119: Send ISDN Progress Parameters
Parameter
Description
Send All
Send an ISDN Progress message in all situations where call progression is
signaled.
Send
Inband
Send an ISDN Progress message only when call progression contains an
indication of in-band information.
Send
Alerting
Send an ISDN Alerting message instead of ISDN Progress message when call
progression contains an indication of in-band information. If call progression does
not contain in-band information, no message is sent at this step.
The strategy for sending Progress messages should be adapted to the configuration of the peer
ISDN switch. Some switches may terminate calls when receiving one or many ISDN progress
messages.
31.
Select the strategy for sending the Progress Indicator Information Element in the Send Progress
Indicator IE drop-down menu.
Table 120: Send Progress Indicator IE Parameters
Parameter
Description
Send All
Send the Progress Indicator IE in all situations.
Send
Inband
Only
Send the Progress Indicator only when the Progress Description contains an
indication of in-band information.
The strategy for the Progress Indicator IE should be adapted to the configuration of the peer ISDN
switch.
This parameter controls sending of a Progress Indicator IE in ISDN messages where Progress
Indicators are allowed. See the parameters in “Interop Parameters Configuration” on page 178 for
a control over which ISDN message allows Progress Indicators.
32.
Select the default value to insert in the "Type of Number" parameter of the "Calling Party Number"
IE when "Type of Number" is not already defined in the call properties in the Default TON for Calling
Party Number IE drop-down menu.
Note: A call property set by the "Properties Manipulation" feature of the Call Router has precedence over
this default value. See “Chapter 44 - Call Router Configuration” on page 431 for more details.
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This parameter applies to the outgoing ISDN calls. Possible values are:
Table 121: Default TON for Calling Party Number IE Parameters
Parameter
Unknown
Description
Default value of Type of Number is set to Unknown.
International Default value of Type of Number is set to International.
33.
National
Default value of Type of Number is set to National.
Network
Specific
Default value of Type of Number is set to Network-Specific.
Subscriber
Default value of Type of Number is set to Subscriber.
Abbreviated
Default value of Type of Number is set to Abbreviated.
Select the default value to insert in the "Numbering Plan Identification" parameter of the "Calling
Party Number" IE when "Numbering Plan Identification" is not already defined in the call properties
in the Default NPI for Calling Party Number IE drop-down menu.
Note: A call property set by the "Properties Manipulation" feature of the Call Router has precedence over
this default value. See “Chapter 44 - Call Router Configuration” on page 431 for more details.
This parameter applies to the outgoing ISDN calls. Possible values are:
Table 122: Default NPI for Calling Party Number IE Parameters
Parameter
34.
Description
Unknown
Default value of Numbering Plan Identification is set to Unknown.
ISDN
Telephony
Default value of Numbering Plan Identification is set to ISDN Telephony.
Data
Default value of Numbering Plan Identification is set to Data.
Telex
Default value of Numbering Plan Identification is set to Telex.
National
Standard
Default value of Numbering Plan Identification is set to National Standard.
Private
Default value of Numbering Plan Identification is set to Private.
Select the default value to insert in the "Presentation Indicator" parameter of the "Calling Party
Number" IE when "Presentation Indicator" is not already defined in the call properties in the Default
PI for Calling Party Number IE drop-down menu.
Note: A call property set by the "Properties Manipulation" feature of the Call Router has precedence over
this default value. See “Chapter 44 - Call Router Configuration” on page 431 for more details.
If "Presentation Indicator" is not provided by the call properties, its value is determined by the
following two steps.
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a.
First, it is set to the default value defined by "DefaultCallingPi".
b.
Second, the "Presentation Indicator" can be overridden by the CLIP and CLIR services: the
value can be set to "Restricted" by the CLIR service and the value can be set to "NotAvailable"
if there is no number to forward. This variable applies to the outgoing ISDN calls.
Possible values are:
Table 123: Default PI for Calling Party Number IE Parameters
Parameter
Description
Presentation
Allowed
Default value of Presentation Indicator is set to Presentation Allowed.
Presentation
Restricted
Default value of Presentation Indicator is set to Presentation Restricted.
Not Available Default value of Presentation Indicator is set to Not Available.
35.
Select the default value to insert in the "Screening Indicator" parameter of the "Calling Party
Number" IE when "Screening Indicator" is not already defined in the call properties in the Default SI
for Calling Party Number IE drop-down menu.
Note: A call property set by the "Properties Manipulation" feature of the Call Router has precedence over
this default value. See “Chapter 44 - Call Router Configuration” on page 431 for more details.
This parameter applies to the outgoing ISDN calls. Possible values are:
Table 124: Default SI for Calling Party Number IE Parameters
Parameter
Description
User Provided Not Screened Default value of Screening Indicator is set to User Provided Not
Screened.
36.
User Provided Verified And
Passed
Default value of Screening Indicator is set to User Provided
Verified And Passed.
User Provided Verified And
Failed
Default value of Screening Indicator is set to User Provided
Verified And Failed.
Network Provided
Default value of Screening Indicator is set to Network Provided.
Context Dependent
Screening Indicator is set to the value that makes the most sense
according to run-time context.
Select the default value to insert in the "Type of Number" parameter of the "Called Party Number"
IE when "Type of Number" is not already defined in the call properties in the Default TON for Called
Party Number IE drop-down menu.
Note: A call property set by the "Properties Manipulation" feature of the Call Router has precedence over
this default value. See “Chapter 44 - Call Router Configuration” on page 431 for more details.
This parameter applies to the outgoing ISDN calls. Possible values are:
Table 125: Default TON for Called Party Number IE Parameters
Parameter
Unknown
Description
Default value of Type of Number is set to Unknown.
International Default value of Type of Number is set to International.
National
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Table 125: Default TON for Called Party Number IE Parameters (Continued)
Parameter
37.
Description
Network
Specific
Default value of Type of Number is set to Network-Specific.
Subscriber
Default value of Type of Number is set to Subscriber.
Abbreviated
Default value of Type of Number is set to Abbreviated.
Select the default value to insert in the "Numbering Plan Identification" parameter of the "Called
Party Number" IE when "Numbering Plan Identification" is not already defined in the call properties
in the Default NPI for Called Party Number IE drop-down menu.
Note: A call property set by the "Properties Manipulation" feature of the Call Router has precedence over
this default value. See “Chapter 44 - Call Router Configuration” on page 431 for more details.
This parameter applies to the outgoing ISDN calls. Possible values are:
Table 126: Default NPI for Called Party Number IE Parameters
Parameter
38.
Description
Unknown
Default value of Numbering Plan Identification is set to Unknown.
ISDN
Telephony
Default value of Numbering Plan Identification is set to ISDN Telephony.
Data
Default value of Numbering Plan Identification is set to Data.
Telex
Default value of Numbering Plan Identification is set to Telex.
National
Standard
Default value of Numbering Plan Identification is set to National Standard.
Private
Default value of Numbering Plan Identification is set to Private.
Select the unit's behaviour when it receives a Notification Indicator IE with the description set to
User suspended in the Notification User Suspended drop-down menu.
Possible values are:
Table 127: Notification User Suspended Parameters
Parameter
39.
Description
Ignore
The Mediatrix unit ignores the Notification Indicator IE with description set to
User suspended.
Disconnect
The Mediatrix unit disconnects the call on Notification Indicator IE with
description set to User suspended.
Click Submit if you do not need to set other parameters.
InformationFollowing Operation
The "informationFollowing" operation is supported for NI2 signaling only (see “PRI Configuration” on page 155
for more details).
When a SETUP message is received containing an "informationFollowing" operation, the unit immediately
sends a PROCEEDING message. The unit then waits normally for a FACILITY message containing the calling
party name, for a maximum time configured with the Maximum Facility Waiting Delay parameter (see “Interop
Parameters Configuration” on page 178 for more details).
The only difference between this behaviour and the usual behaviour (i.e. without the "informationFollowing"
operation), is the immediate sending of the PROCEEDING message before waiting for the calling party name.
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Note that the "informationFollowing" operation is mutually exclusive with the Call Proceeding Delay parameter
(see “Interop Parameters Configuration” on page 178 for more details), which configures a delay before
sending the PROCEEDING message. If the PROCEEDING message is sent due to the "informationFollowing"
operation, Call Proceeding Delay parameter is ignored.
BRI Configuration
This section applies to the following models:






Mediatrix 3404
Mediatrix 3408
Mediatrix 3734
Mediatrix 3741
Mediatrix 3742
Mediatrix 4400 Series
A Basic Rate Interface (BRI) port supports 2 x 64 kbit/s B-channels for switched voice or data connections and
1 x 16 kbit/s D-channel for signalling.
Caution: The Mediatrix unit ISDN BRI ports are configurable to operate as network or terminal ports. The
pin-out of the sockets is switched according to this configuration. Wrong port configurations, wrong cabling
or wrong connections to neighbouring equipment can lead to short circuits in the BRI line powering. Refer
to the Hardware Installation Guide to avoid misconfigurations.
Caution: The Mediatrix unit BRI ports can be used as a S or T reference point, but not as U reference points
(2-wire). Never connect a U SCN line or a U TE into the Mediatrix unit BRI ports.
The Mediatrix 3404 / 3734 / 3741 / 3742 has 5 ISDN BRI ports. It supports up to 8 simultaneous ISDN voice/
data channels over any IP connection.
The Mediatrix 3408 has 10 ISDN BRI ports. It supports up to 16 simultaneous ISDN voice/data channels over
any IP connection.
The Mediatrix 4400 Series has up to 4 ISDN BRI ports depending on the model. It supports up to 8
simultaneous ISDN voice/data channels over any IP connection.
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 To configure the BRI parameters:
1.
In the web interface, click the ISDN link, then the Basic Rate Interface sub-link.
Figure 67: ISDN – Basic Rate Interface Web Page
2
3
5
4
6
7
8
9
10
11
12
13
15
17
19
21
23
25
14
16
18
20
22
24
26
27
28
29
31
30
32
33
34
2
2.
In the Interface Configuration section of the Basic Rate Interface page, select to which interface you
want to apply the changes in the Select Interface drop-down menu at the top of the window.
You can copy the configuration of the selected interface to one or more interfaces of the Mediatrix
unit in the Apply to the Following Interfaces section at the bottom of the page. You can select
specific interfaces by checking them, as well as use the Check All or Uncheck All buttons.
The Mediatrix 3404 model has 5 interfaces in Slot 1, while the Mediatrix 3408 model has 10
interfaces in Slots 1 and 2 (5 in each).
3.
Select the endpoint type in the Endpoint Type drop-down menu.
Table 128: Endpoint Type
Type
168
Description
TE
Terminal Equipment.The endpoint emulates the subscriber (terminal) side of the digital
connection. You can connect the SCN to the endpoint.
NT
Network Termination. The endpoint emulates the central office (network) side of the
digital connection. You can connect a PBX or ISDN telephones to the endpoint.
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The setting used for the Mediatrix unit must be opposite to the setting used in the PBX. For instance,
if the PBX is set to TE, then the Mediatrix unit must be set to NT.
Note: If you want to use a specific interface as the reference clock, you must set it to TE.
When the BRI interface Signalling Protocol drop-down is set to QSIG (see Step 7), the endpoint type
is only used in the second layer (LAPD) since it is a concept that does not exist in QSIG.
4.
Select the clock mode of the interface in the Clock Mode drop-down menu.
The interface can either generate the clocking for the line or accept the clock from the line.
Table 129: Clock Mode
Mode
auto
Description
The setting is derived from the endpoint type.
•
NT: clock master
•
TE: clock slave
Master The interface generates the clock.
Slave
The interface accepts the clock from the line.
The clock mode is used to give the user the possibility to set an endpoint in TE mode and still
generate the clock by specifying the clock mode to master. The clock source can then be selected
from the Clock Reference drop-down menu (see “Chapter 5 - Hardware Parameters” on page 29
for more details). The clock mode could be used, for instance, to synchronize several units in NT
mode via a BRI line.
Note: In a BRI configuration, setting the clock mode to master for a TE endpoint is invalid. Slave mode is
automatically applied in this case.
5.
Set the Monitor Link State drop-down menu with the physical link state of the ISDN interface.
Table 130: Interface Link State
Parameter
Description
Enable
The ISDN endpoint's operational state is affected by its interface physical link state.
When the link state of the ISDN interface is down, the operational state of its
matching endpoint becomes “disable”.
Disable
The ISDN endpoint's operational state is not affected by its interface physical link
state.
Note that if the Monitor Link State parameter is enabled and the Ignore SIP OPTONS on no usable
endpoints parameter is also enabled in the SIP / Interop page, this will influence how the SIP options
are answered. See “SIP Interop” on page 279 for more details.
6.
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Select the connection type of the endpoint in the Connection Type drop-down menu.
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The type of connection depends on the equipment to which the Mediatrix unit port is connected and
it must be the same for all interconnected pieces of equipment.
Table 131: Connection Type
Type
Description
Point to Point
Can only attach one device (for instance a PBX or SCN) and acts as a T
reference point.
Point to
MultiPoint
Can attach more than one ISDN device and acts as a S reference point. Up
to 8 TEs and one NT can be connected to a S-bus.
Note: If you are using a Mediatrix unit connected to a S-Bus in point-to-multipoint TE mode, you cannot
currently connect any additional ISDN devices to the S-Bus.
The Point to MultiPoint configuration is not available in QSIG.
7.
Select the protocol to use for the signalling channel in the Signalling Protocol drop-down menu.
This signalling must match the connected ISDN equipment or network.
Table 132: Signalling Protocols
Protocol
Description
DSS1
Digital Subscriber Signaling System No.1
DMS100
Digital Multiplex System 100
NI2
National ISDN No.2
5ESS
5 Electronic Switching System
QSIG
ECMA's protocol for Private Integrated
Services Networks
Note: The Dgw v2.0 Application currently supports only the DSS1 and QSIG signalling protocols.
8.
Select the value of the network location in the progress indicator messages that the unit sends in
the Network Location drop-down menu.
This defines the location code to be inserted in ISDN causes code information elements, i.e., the
type of network to which the system belongs. The following values are available:
9.
•
User
•
Private
•
Public
•
Transit
•
International
Set the Preferred Encoding Scheme drop-down menu with the data encoding scheme in the bearer
capabilities (user information layer 1 protocol).
This encoding scheme is used when initiating a call on the ISDN side. The supported encoding
schemes are G.711 u-Law and G.711 a-Law.
G.711 u-Law may not be supported by DSS1 NT and TE endpoints. It is recommended to use G.711
a-Law as preferred encoding protocol.
10.
170
Set the Fallback Encoding Scheme drop-down menu with the fallback data encoding scheme in
case the preferred encoding scheme is not available.
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The supported encoding schemes are G.711 u-Law and G.711 a-Law.
Note: The fallback encoding scheme is valid only when receiving a SETUP message. The user sending the
SETUP message does not indicate alternative bearer capability.
If the proposed encoding scheme in the bearer capability received in the SETUP message is
different than the preferred encoding scheme, then the fallback encoding scheme is used.
11.
Select the strategy for selecting bearer channels in the Channel Allocation Strategy drop-down
menu.
Table 133: Channel Allocation Strategy
Allocation
Description
Ascending
Starting from the lowest-numbered non-busy bearer channel and
going toward the highest-numbered non-busy bearer channel, the
Mediatrix unit selects the first bearer channel available.
Descending
Starting from the highest-numbered non-busy bearer channel and
going toward the lowest-numbered non-busy bearer channel, the
Mediatrix unit selects the first bearer channel available.
RoundRobinAscending
The Mediatrix unit starts from the bearer channel that follows the
bearer channel used for the last call. For instance, if channel #1 was
used in the last call, the unit starts with channel #2. Going toward the
highest-numbered non-busy bearer channel, the unit selects the first
channel available. If the highest channel is unavailable, the search
continues from the lowest-numbered non-busy bearer channel.
RoundRobinDescending The Mediatrix unit starts from the bearer channel that precedes the
bearer channel used for the last call. For instance, if channel #3 was
used in the last call, the unit starts with channel #2. Going toward the
lowest-numbered non-busy bearer channel, the unit selects the first
channel available. If the lowest channel is unavailable, the search
continues from the highest-numbered non-busy bearer channel.
12.
Define the maximum number of active calls on the interface in the Maximum Active Calls field.
This limits the total number of concurrent calls on the interface. Entering 0 indicates no maximum
number of active calls.
For a Mediatrix 3404 / 3408, the maximum number of simultaneous calls is limited to 8, even if 5
BRI ports can physically support 10 calls.
13.
14.
Select whether or not the signal information element is enabled in the Signal Information Element
drop-down menu.
•
When activated at the Network UNI-side (Endpoint Type drop-down menu set to NT),
the signal information element is sent to the User UNI-side.
•
When activated at the User UNI-side (Endpoint Type drop-down menu set to TE), the
tone indicated in the signal information element is played in the IP direction.
Select whether or not inband tone generation is enabled in the Inband Tone Generation drop-down
menu.
When activated at the User UNI-side (Endpoint Type drop-down menu is set to TE), only the
ringback tone is generated.
When the BRI interface Signalling Protocol drop-down is set to QSIG (see Step 7) and this
parameter is activated, the incoming side of the call plays the tones inband.
15.
Select whether or not inband DTMF dialing is enabled in the Inband DTMF Dialing drop-down menu.
If you select Enable, the Mediatrix unit accepts inband DTMF digits when Overlap Dialing occurs.
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16.
BRI Configuration
Select whether or not overlap dialing is enabled in the Overlap Dialing drop-down menu.
Table 134: Overlap Dialing Parameters
Parameter
17.
Description
Enable
The Mediatrix unit transports the called-party number digit by digit, after the first
SETUP message, which contains no called party information at all.
Disable
The Mediatrix unit transports the full called party information in the first SETUP
message from the terminal. This means that the user must dial the number before
going off-hook.
Define the maximum length of the calling party name for calls from SIP to ISDN in the Calling Name
Max Length field.
Available values range from 0 to 82.
18.
Select whether or not exclusive B-Channel selection is enabled for calls from SIP to ISDN in the
Exclusive B-Channel Selection drop-down menu.
If you select Enable and initiate a call, only the requested B channel is accepted; if the requested
B channel is not available, the call is cleared.
19.
Select whether or not to enable the Sending Complete information element into SETUP messages
for calls from SIP to ISDN in the Sending Complete drop-down menu.
Some ISDN switches may require that the Sending Complete information element be included in
the outgoing SETUP message to indicate that the entire number is included and there are no further
destination digits to be sent.
20.
Select whether or not to enable sending the RESTART message upon a signalling channel “UP”
event in the Send Restart On Startup drop-down menu.
The RESTART message requests a restart for the interface specified.
21.
Set the link establishment strategy in the Link Establishment drop-down menu.
Table 135: Link Establishment Parameters
Parameter
Description
OnDemand When the data link is shut down, the unit establishes a new link only when
required.
Permanent
22.
When the data link is shut down, the unit immediately attempts to establish a new
link.
Set the actual keypad string that is to be considered as a hook-flash in the Hook-Flash Keypad field.
An ISDN telephone may send INFORMATION messages that contain a “Keypad Facility”. You can
thus trigger a supplementary service (Hold, Conference, etc.) by sending a keypad facility.
Since the keypads can be received via several INFORMATION messages, they are accumulated
until they match or reset if the keypad reception timeout (second) has elapsed since the last keypad
has been received. The keypad reception timeout can only be modified via SNMP. If the keypad
reception timeout is set to 0, it disables the timeout, thus assuming that all keypads will be received
in a single INFORMATION message.
Setting this variable to an empty string disables the hook-flash detection.
The permitted keypad must be made up of IA5 characters. See ITU-T Recommendation T.50.
23.
Set the STATUS causes that can be received without automatically clearing the call in the Accepted
Status Causes field.
The default action is to clear the call upon receiving a STATUS message. If a STATUS message is
received indicating a compatible call state and containing the supplied STATUS causes, the
clearing of the call is prevented.
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The string has the following syntax:
•
',': Separator between non-consecutive lists of causes or single cause.</li>
•
'n': A single cause, where n is the cause number.</li>
'm-n': List of causes where m is the start cause number and n is the end cause number.
Note: The space character is ignored and cause duplication is not allowed.
Causes must be specified in low to high order.
Example: '1,124-127': The accepted causes are 1, 124, 125, 126 and 127.
24.
Set the range of PROGRESS causes accepted by the unit in the Accepted Progress Causes field.
Causes excluded from this range trigger call disconnections.
The string has the following syntax:
•
',': Separator between non-consecutive lists of causes or single cause.
•
'n': A single cause, where n is the cause number.
•
'm-n': List of causes where m is the start cause number and n is the end cause number.
Note: You must consider the following:
• The space character is not allowed.
• Causes must be specified in low to high order.
• Cause duplication is not allowed.
Example: '1,124-127': The accepted causes are 1, 124, 125, 126 and 127.
25.
Select the strategy for sending ISDN Progress messages in the Send Isdn Progress drop-down
menu.
Table 136: Send ISDN Progress Parameters
Parameter
Description
Send All
Send an ISDN Progress message in all situations where call progression is
signaled.
Send
Inband
Send an ISDN Progress message only when call progression contains an
indication of in-band information.
Send
Alerting
Send an ISDN Alerting message instead of ISDN Progress message when call
progression contains an indication of in-band information. If call progression does
not contain in-band information, no message is sent at this step.
The strategy for sending Progress messages should be adapted to the configuration of the peer
ISDN switch. Some switches may terminate calls when receiving one or many ISDN progress
messages.
26.
Select the strategy for sending the Progress Indicator Information Element in the Send Progress
Indicator IE drop-down menu.
Table 137: Send Progress Indicator IE Parameters
Parameter
Description
Send All
Send the Progress Indicator IE in all situations.
Send
Inband
Only
Send the Progress Indicator only when the Progress Description contains an
indication of in-band information.
The strategy for the Progress Indicator IE should be adapted to the configuration of the peer ISDN
switch.
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This parameter controls sending of a Progress Indicator IE in ISDN messages where Progress
Indicators are allowed. See the parameters in “Interop Parameters Configuration” on page 178 for
a control over which ISDN message allows Progress Indicators.
27.
Set the TEI Negotiation drop-down menu with the proper Terminal Endpoint Identifier (TEI)
negotiation strategy.
Table 138: TEI Negotiation Parameters
Parameter
Description
Link Up
Each time the physical link comes up, the unit renegotiates the TEI value.
Power Up
When the physical link comes up, the unit does not renegotiate the TEI value. The
value obtained at power-up is reused.
Signaling
up
Each time the signaling link comes up, the unit renegotiates the TEI value.
Note: This parameter only applies on Point To Multipoint connections.
28.
Select the default value to insert in the "Type of Number" parameter of the "Calling Party Number"
IE when "Type of Number" is not already defined in the call properties in the Default TON for Calling
Party Number IE drop-down menu.
Note: A call property set by the "Properties Manipulation" feature of the Call Router has precedence over
this default value. See “Chapter 44 - Call Router Configuration” on page 431 for more details.
This parameter applies to the outgoing ISDN calls. Possible values are:
Table 139: Default TON for Calling Party Number IE Parameters
Parameter
Unknown
Description
Default value of Type of Number is set to Unknown.
International Default value of Type of Number is set to International.
29.
National
Default value of Type of Number is set to National.
Network
Specific
Default value of Type of Number is set to Network-Specific.
Subscriber
Default value of Type of Number is set to Subscriber.
Abbreviated
Default value of Type of Number is set to Abbreviated.
Select the default value to insert in the "Numbering Plan Identification" parameter of the "Calling
Party Number" IE when "Numbering Plan Identification" is not already defined in the call properties
in the Default NPI for Calling Party Number IE drop-down menu.
Note: A call property set by the "Properties Manipulation" feature of the Call Router has precedence over
this default value. See “Chapter 44 - Call Router Configuration” on page 431 for more details.
This parameter applies to the outgoing ISDN calls. Possible values are:
Table 140: Default NPI for Calling Party Number IE Parameters
Parameter
174
Description
Unknown
Default value of Numbering Plan Identification is set to Unknown.
ISDN
Telephony
Default value of Numbering Plan Identification is set to ISDN Telephony.
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Software Configuration Guide
Table 140: Default NPI for Calling Party Number IE Parameters (Continued)
Parameter
30.
Description
Data
Default value of Numbering Plan Identification is set to Data.
Telex
Default value of Numbering Plan Identification is set to Telex.
National
Standard
Default value of Numbering Plan Identification is set to National Standard.
Private
Default value of Numbering Plan Identification is set to Private.
Select the default value to insert in the "Presentation Indicator" parameter of the "Calling Party
Number" IE when "Presentation Indicator" is not already defined in the call properties in the Default
PI for Calling Party Number IE drop-down menu.
Note: A call property set by the "Properties Manipulation" feature of the Call Router has precedence over
this default value. See “Chapter 44 - Call Router Configuration” on page 431 for more details.
If "Presentation Indicator" is not provided by the call properties, its value is determined by the
following two steps.
a.
First, it is set to the default value defined by "DefaultCallingPi".
b.
Second, the "Presentation Indicator" can be overridden by the CLIP and CLIR services: the
value can be set to "Restricted" by the CLIR service and the value can be set to "NotAvailable"
if there is no number to forward. This variable applies to the outgoing ISDN calls.
Possible values are:
Table 141: Default PI for Calling Party Number IE Parameters
Parameter
Description
Presentation
Allowed
Default value of Presentation Indicator is set to Presentation Allowed.
Presentation
Restricted
Default value of Presentation Indicator is set to Presentation Restricted.
Not Available Default value of Presentation Indicator is set to Not Available.
31.
Select the default value to insert in the "Screening Indicator" parameter of the "Calling Party
Number" IE when "Screening Indicator" is not already defined in the call properties in the Default SI
for Calling Party Number IE drop-down menu.
Note: A call property set by the "Properties Manipulation" feature of the Call Router has precedence over
this default value. See “Chapter 44 - Call Router Configuration” on page 431 for more details.
This parameter applies to the outgoing ISDN calls. Possible values are:
Table 142: Default SI for Calling Party Number IE Parameters
Parameter
Description
User Provided Not Screened Default value of Screening Indicator is set to User Provided Not
Screened.
Dgw v2.0 Application
User Provided Verified And
Passed
Default value of Screening Indicator is set to User Provided
Verified And Passed.
User Provided Verified And
Failed
Default value of Screening Indicator is set to User Provided
Verified And Failed.
Network Provided
Default value of Screening Indicator is set to Network Provided.
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Chapter 23 - ISDN Configuration
BRI Configuration
Table 142: Default SI for Calling Party Number IE Parameters (Continued)
Parameter
Context Dependent
32.
Description
Screening Indicator is set to the value that makes the most sense
according to run-time context.
Select the default value to insert in the "Type of Number" parameter of the "Called Party Number"
IE when "Type of Number" is not already defined in the call properties in the Default TON for Called
Party Number IE drop-down menu.
Note: A call property set by the "Properties Manipulation" feature of the Call Router has precedence over
this default value. See “Chapter 44 - Call Router Configuration” on page 431 for more details.
This parameter applies to the outgoing ISDN calls. Possible values are:
Table 143: Default TON for Called Party Number IE Parameters
Parameter
Unknown
Description
Default value of Type of Number is set to Unknown.
International Default value of Type of Number is set to International.
33.
National
Default value of Type of Number is set to National.
Network
Specific
Default value of Type of Number is set to Network-Specific.
Subscriber
Default value of Type of Number is set to Subscriber.
Abbreviated
Default value of Type of Number is set to Abbreviated.
Select the default value to insert in the "Numbering Plan Identification" parameter of the "Called
Party Number" IE when "Numbering Plan Identification" is not already defined in the call properties
in the Default NPI for Called Party Number IE drop-down menu.
Note: A call property set by the "Properties Manipulation" feature of the Call Router has precedence over
this default value. See “Chapter 44 - Call Router Configuration” on page 431 for more details.
This parameter applies to the outgoing ISDN calls. Possible values are:
Table 144: Default NPI for Called Party Number IE Parameters
Parameter
34.
176
Description
Unknown
Default value of Numbering Plan Identification is set to Unknown.
ISDN
Telephony
Default value of Numbering Plan Identification is set to ISDN Telephony.
Data
Default value of Numbering Plan Identification is set to Data.
Telex
Default value of Numbering Plan Identification is set to Telex.
National
Standard
Default value of Numbering Plan Identification is set to National Standard.
Private
Default value of Numbering Plan Identification is set to Private.
Select the unit's behaviour when it receives a Notification Indicator IE with the description set to
User suspended in the Notification User Suspended drop-down menu.
Dgw v2.0 Application
BRI Configuration
Software Configuration Guide
Possible values are:
Table 145: Notification User Suspended Parameters
Parameter
35.
Description
Ignore
The Mediatrix unit ignores the Notification Indicator IE with description set to
User suspended.
Disconnect
The Mediatrix unit disconnects the call on Notification Indicator IE with
description set to User suspended.
Click Submit if you do not need to set other parameters.
Bypass Feature (Mediatrix 3404/3408/3734/3741/3742 Models)
In the event of a power or network failure, the bypass feature permits users to make and receive calls even
when the Mediatrix unit is not operating. The Mediatrix unit BRI 3 and BRI 4 ports may either act as a SCN
bypass. For instance, if you decide to connect a SCN line into the BRI 4 port, you can use a BRI telephone
connected into the BRI 3 port to make calls.
Furthermore:


The port on which the SCN line is connected must be configured as a TE.
The other port must be configured as a NT.
Refer to “BRI Configuration” on page 167 for more details on how to configure the line type.
During normal operation, the direct connection between the BRI 3 and BRI 4 ports is switched out through
commuting relays and both ports resume normal functions. When power is removed from the Mediatrix unit,
the relay setting is restored to a connected state and the SCN line can be used as an emergency line.
Consequently, a BRI telephone used on the other port is directly connected to this SCN line. When the power
is restored, this automatically removes the Bypass connection; this means that any ongoing call on the Bypass
connection is terminated.
Note: If you are using a crossover Ethernet cable to connect the SCN line to the Mediatrix unit and there is
a power failure, the bypass feature does not work properly.
Bypass Feature (Mediatrix 4402plus / 4404plus Models)
In the event of a power or network failure, the optional bypass feature permits users to make and receive calls
even when the Mediatrix unit is not operating. The Mediatrix unit BRI 1 and BRI 2 ports may either act as a
SCN bypass. For instance, if you decide to connect a SCN line into the BRI 2 port, you can use a BRI
telephone connected into the BRI 1 port to make calls.
Furthermore:


The port on which the SCN line is connected must be configured as a TE.
The other port must be configured as a NT.
Refer to “BRI Configuration” on page 167 for more details on how to configure the line type.
During normal operation, the direct connection between the BRI 1 and BRI 2 ports is switched out through
commuting relays and both ports resume normal functions. When power is removed from the Mediatrix unit,
the relay setting is restored to a connected state and the SCN line can be used as an emergency line.
Consequently, a BRI telephone used on the other port is directly connected to this SCN line. When the power
is restored, this automatically removes the Bypass connection; this means that any ongoing call on the Bypass
connection is terminated.
Dgw v2.0 Application
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Chapter 23 - ISDN Configuration
Interop Parameters Configuration
Interop Parameters Configuration
The interop parameters allow the Mediatrix unit to properly work, communicate, or connect with specific ISDN
devices.
 To set the interop parameters:
1.
In the web interface, click the ISDN link, then the Interop sub-link.
Figure 68: ISDN – Interop Web Page
2
3
4
5
7
9
11
6
8
10
12
13
2
2.
Select to which interface you want to apply the changes in the Select Interface drop-down menu at
the top of the window.
You can copy the configuration of the selected interface to one or more interfaces of the Mediatrix
unit in the Apply to the Following Interfaces section at the bottom of the page. You can select
specific interfaces by checking them, as well as use the Check All or Uncheck All buttons.
The Mediatrix 3404 model has 5 interfaces in Slot 2, while the Mediatrix 3408 model has 10
interfaces in Slots 2 and 3 (5 in each).
The Mediatrix 3532 and 3632 models have two interfaces.
The Mediatrix 3734/3741/3742 models have 5 interfaces.
The number of interfaces available vary depending on the Mediatrix 4400 model you have.
3.
Define the behaviour of the Progress Indicator In Setup drop-down menu.
This menu defines whether or not the Progress Indicator Information Element (IE) is allowed in the
SETUP message when acting as the originating side.
See the Send Progress Indicator IE parameter for other conditions for sending Progress Indicator
IE (“PRI Configuration” on page 155 and “BRI Configuration” on page 167).
4.
Define the behaviour of the Progress Indicator In Setup Ack drop-down menu.
This menu defines whether or not the Progress Indicator Information Element (IE) is allowed in the
SETUP ACK when acting as the terminating side.
See the Send Progress Indicator IE parameter for other conditions for sending Progress Indicator
IE (“PRI Configuration” on page 155 and “BRI Configuration” on page 167).
5.
Define the behaviour of the Progress Indicator In Call Proceeding drop-down menu.
Thie menu defines whether or not the Progress Indicator Information Element (IE) is allowed in the
CALL PROCEEDING message in response to a SETUP message when acting as the terminating
side.
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See the Send Progress Indicator IE parameter for other conditions for sending Progress Indicator
IE (“PRI Configuration” on page 155 and “BRI Configuration” on page 167).
6.
Define the behaviour of the Progress Indicator In Progress drop-down menu.
This menu defines whether or not the Progress Indicator Information Element (IE) is allowed in the
PROGRESS message in response to a SETUP message when acting as the terminating side.
See the Send Progress Indicator IE parameter for other conditions for sending Progress Indicator
IE (“PRI Configuration” on page 155 and “BRI Configuration” on page 167).
7.
Define the behaviour of the Progress Indicator In Alerting drop-down menu.
This menu defines whether or not the Progress Indicator Information Element (IE) is allowed in the
ALERTING message.
See the Send Progress Indicator IE parameter for other conditions for sending Progress Indicator
IE (“PRI Configuration” on page 155 and “BRI Configuration” on page 167).
8.
Define the behaviour of the Progress Indicator In Connect drop-down menu.
This menu defines whether or not the Progress Indicator Information Element (IE) is allowed in the
CONNECT message.
See the Send Progress Indicator IE parameter for other conditions for sending Progress Indicator
IE (“PRI Configuration” on page 155 and “BRI Configuration” on page 167).
9.
Define a value, in milliseconds (ms), in the Maximum Facility Waiting Delay (ms) field.
This value defines the maximum amount of time to wait for a FACILITY message, after receiving a
SETUP message, before going on with normal call processing.
After receiving a SETUP message, the system waits for this amount of time for a FACILITY
message. As soon as it receives a FACILITY message or the delay expires, it goes on with normal
call processing.
A FACILITY message can contain useful information for the call. For example, it can contain a
Calling Name.
You must enable the Supplementary Services to use the delay. See “PRI Configuration” on
page 155 or “BRI Configuration” on page 167 for more details.
Setting the value to 0 deactivates this waiting delay.
10.
Define whether or not a message with progress indicator No. 1 MUST be considered as offering
inband information available in the Use Implicit Inband Info drop-down menu.
If so, the network must activate the B-channel connection.
The progress indicator No. 1 means that the call is not end-to-end ISDN; further call progress
information may be available inband.
11.
Defines the maximum time, in milliseconds, to wait after receiving a SETUP message before
sending a CALL PROCEEDING message and going on with normal call processing in the Call
Proceeding Delay field.
After receiving a SETUP message, the system waits for a message from the called party. If the
message maps to a User Busy cause, a DISCONNECT message is sent instead of the CALL
PROCEEDING otherwise it goes on with normal call processing.
The value 0 deactivates this feature.
12.
Define how the Calling Name is delivered in the Calling Name Delivery field.
The Calling Party Name can be received and sent through three different methods: Facility
information element, Display information element or User-User information element.
Table 146: Calling Name Delivery Parameters
Parameter
Dgw v2.0 Application
Description
DisplayIe
Use a Display Information Element for delivering the Calling Name.
FacilityIe
Use a Facility Information Element for delivering the Calling Name.
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Chapter 23 - ISDN Configuration
Interop Parameters Configuration
Table 146: Calling Name Delivery Parameters (Continued)
Parameter
Description
UserUserI
e
Use a User-User Information Element for delivering the Calling Name.
SignalingP
rotocol
Use the delivery method defined by the signaling protocol.
When receiving an incoming call, the three possible sources of Calling Party Name are checked in
the following order: User-User, Display and Facility. The last found is used.
The Calling Party Name is accepted in a Display information element only when explicitly identified
as a Calling Party Name (i.e. only when "Display Type" = "Calling Party Name" in the information
element).
When initiating a call, the Calling Party Name is sent according to the method selected above. If the
method selected is not supported for the protocol in use, the default method for this protocol is used.
The following table shows which method is used vs. the configuration of CallingNameDelivery:
Table 147: Calling Name Delivery Method vs. Configuration
Calling Name Delivery
Protocol
eFacility
eDisplay
eUserUser
eSignalingProtocol
DSS1
IE User-User
IE User-User
IE User-User
IE User-User
Dms100
IE Facility
IE Display
IE Display
IE Display
NI-2
IE Facility
IE Facility
IE Facility
IE Facility
5ESS
IE Facility
IE Facility
IE User-User
IE Facility
QSIG
IE Facility
IE Facility
IE Facility
IE Facility
See “PRI Configuration” on page 155 and “BRI Configuration” on page 167 for more details on
signaling protocols.
13.
Define whether or not an ISDN message with a TEI broadcast needs to be interpreted as a TEI 0
when the connection type is 'PointToPoint' in the Allow TEI Broadcast in PTP drop-down menu.
See “BRI Configuration” on page 167 for more details on the connection type.
This parameter is available for the following BRI models:
14.
•
Mediatrix 3404
•
Mediatrix 3408
•
Mediatrix 3734
•
Mediatrix 3741
•
Mediatrix 3742
•
Mediatrix 4400 Series
Click Submit if you do not need to set other parameters.
Play Local Ringback when no Media Stream
This section describes configuration that is available only in the MIB parameters of the Mediatrix unit. You can
configure these parameters as follows:



by using a MIB browser
by using the CLI
by creating a configuration script containing the configuration variables
You can force the local ringback generation when early-media is enabled but no media stream has been
received yet. This variable only affects incoming calls on the ISDN interface.
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Software Configuration Guide
Note that this variable only applies to 180 SIP responses when early-media is enabled.
The following configurations are supported:
Table 148: Play Local Ringback Configuration
Configuration
Description
disable
Do not play local ringback when doing early-media.
enable
The local ringback is played after sending an ALERTING and no media stream has been
received yet from the outgoing interface.
 To set how to play the local ringback when there is no media stream:
1.
In the isdnMIB, set the Play Local Ringback configuration in the
InteropPlayLocalRingbackWhenNoMediaStream variable.
You can also use the following line in the CLI or a configuration script:
isdn.InteropPlayLocalRingbackWhenNoMediaStream="Value"
where Value may be as follows:
Table 149: Play Local Ringback Values
Value Meaning
Dgw v2.0 Application
0
disable
1
enable
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Chapter 23 - ISDN Configuration
ISDN Timers Configuration
ISDN Timers Configuration
This section allows you to set timer parameters.
 To set the ISDN timers:
1.
In the web interface, click the ISDN link, then the Timer sub-link.
Figure 69: ISDN – Timer Web Page
2
3
4
2
2.
Select to which interface you want to apply the changes in the Select Interface drop-down menu at
the top of the window.
You can copy the configuration of the selected interface to one or more interfaces of the Mediatrix
unit in the Apply to the Following Interfaces section at the bottom of the page. You can select
specific interfaces by checking them, as well as use the Check All or Uncheck All buttons.
The Mediatrix 3404 model has 5 interfaces in Slot 2, while the Mediatrix 3408 model has 10
interfaces in Slots 2 and 3 (5 in each).
The Mediatrix 3532 and 3632 models have two interfaces.
The Mediatrix 3734/3741/3742 models have 5 interfaces.
The number of interfaces available vary depending on the Mediatrix 4400 model you have.
3.
Set the Auto Cancel Timeout field with the time, in seconds, the endpoint rings before the call is
automatically cancelled.
Setting this variable to 0 disables the timeout. Calls will not be automatically cancelled and will ring
until the party answers.
4.
Set the value, in milliseconds (ms), of the Layer 1 Timer T3 in the L1 Timer T3 field.
Timer 3 (T3) is a supervisory timer that has to take into account the overall time to activate. This
time includes the time it takes to activate both the TE-NT and the NT-TE portion of the customer
access.
The expiry of Timer T3 is intended to provide an indication that the network side cannot complete
the activation procedure, probably due to a failure condition or the terminal cannot detect INFO 4.
5.
182
Click Submit if you do not need to set other parameters.
Dgw v2.0 Application
Services Configuration
Software Configuration Guide
Services Configuration
This section allows you to set the ISDN optional services.
Standards Supported
•
ETS 300 207: Call Diversion and Call Rerouting
 To set the ISDN optional services:
1.
In the web interface, click the ISDN link, then the Services sub-link.
Figure 70: ISDN – Services Web Page
2
3
4
5
7
9
6
8
10
11
13
12
14
15
16
2
2.
Select to which interface you want to apply the changes in the Select Interface drop-down menu at
the top of the window.
You can copy the configuration of the selected interface to one or more interfaces of the Mediatrix
unit in the Apply to the Following Interfaces section at the bottom of the page. You can select
specific interfaces by checking them, as well as use the Check All or Uncheck All buttons.
The Mediatrix 3404 model has 5 interfaces in Slot 2, while the Mediatrix 3408 model has 10
interfaces in Slots 2 and 3 (5 in each).
The Mediatrix 3532 and 3632 models have two interfaces.
The Mediatrix 3734/3741/3742 models have 5 interfaces.
The number of interfaces available vary depending on the Mediatrix 4400 model you have.
3.
Select whether or not supplementary services FACILITY messages and FACILITY information
elements should be enabled on the ISDN interface in the Facility Services drop-down menu.
This controls how to use the FACILITY information element.
Table 150: Facility Services
Parameter
Disable
Dgw v2.0 Application
Description
Facility information elements are disabled in both directions (send and receive). No
Facility information element can be inserted in sent messages. When receiving a
FACILITY message containing supplementary services information, the Mediatrix
unit replies with a STATUS message saying the FACILITY is not supported.
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Chapter 23 - ISDN Configuration
Services Configuration
Table 150: Facility Services (Continued)
Parameter
Enable
Description
Facility information elements are enabled in both directions (send and receive).
Facility information elements can be inserted in sent messages.
When receiving a FACILITY message containing supplementary services
information, the Mediatrix unit accepts and interprets the message, processing
supported supplementary service messages and silently discarding unsupported
supplementary service messages.
Generic procedures for the control of ISDN supplementary services are defined in the
recommendation ITU-T Q.932.
You can define a waiting delay as described in the section “Interop Parameters Configuration” on
page 178.
Note: To activate the ISDN hold feature, you must also set the Default Hook Flash Processing feature to
Use Signaling Protocol in “General Configuration” on page 385.
4.
Select whether or not to enable the CLIP service in the Calling Line Information Presentation dropdown menu.
The Calling Line Information Presentation (CLIP) is an optional service that can be offered by the
ISDN provider. CLIP is offered to the called party to see the calling party's ISDN number. CLIP is
complemented by privacy rules controlled by the Calling Line Information Restriction and Calling
Line Information Restriction Override parameters.
The Calling Line Information Presentation parameter has the following effect:
Parameter
5.
Description
UserOnly
Sends a Calling Number IE that contains the calling number digits when acting as
the User UNI-side (Endpoint Type drop-down menu set to TE) otherwise the
Calling Number IE is not sent.
Enable
Sends a Calling Number IE that contains the calling number digits.
Disable
The Calling Number IE is never sent.
Select whether or not to enable the CLIR service in the Calling Line Information Restriction dropdown menu.
The Calling Line Information Restriction (CLIR) service is offered to the calling party to restrict
presentation of the calling party's ISDN number to the called party.
Setting this parameter to Disable disables the CLIR service.
Setting this parameter to Enable enables the CLIR service. This has the following effects:
For all ISDN signaling protocols except QSIG:
•
On a TE interface (Endpoint Type drop-down menu set to TE) at the originating network
side, when sending a SETUP message with a Calling Party Number (CPN) IE, the
Presentation Indicator (PI) is set to "Restricted". The calling party number itself is
included in the CPN IE if available.
•
On a NT interface (Endpoint Type drop-down menu set to NT) at the originating
network side, when receiving a SETUP message with a CPN IE, the PI is set to
"Restricted". The calling party number itself is forwarded.
For the QSIG signaling protocol (PRI/BRI interface Signalling Protocol drop-down is set to QSIG):
•
184
Sending a SETUP message: The PI is set to "Restricted" in the CPN IE inserted in the
SETUP message sent to the ISDN, unless the CLIR override option is set. However,
even if PI is set to "Restricted", the calling number is included in the CPN IE.
Dgw v2.0 Application
Services Configuration
Software Configuration Guide
•
Receiving a SETUP message: If PI is set in the received message, the calling party
number is removed, unless the CLIR override option is set.
See “PRI Configuration” on page 155 or “BRI Configuration” on page 167 for more details.
See “PRI Configuration” on page 155 or “BRI Configuration” on page 167 for more details.
6.
Select whether or not to enable the CLIR override option in the Calling Line Information Restriction
Override drop-down menu.
This option allows the calling party number to be presented to the destination party even when the
Calling Party Number (CPN) IE's Presentation Indicator (PI) is set to "Restricted". This option is
typically used for police or emergency services.
Setting this variable to Disable disables the CLIR Override option.
Setting this variable to Enable enables the CLIR Override option. This has the following effects:
For all ISDN signaling protocols except QSIG:
•
The override option acts on the NT interface of the destination network side. It prevents
the number to be removed from the CPN IE inserted in the SETUP message sent to
the destination TE.
For the QSIG signaling protocol (PRI/BRI interface Signalling Protocol drop-down is set to QSIG):
•
The override option prevents the calling name to be removed from the CPN IE in a
received SETUP message.
See “PRI Configuration” on page 155 or “BRI Configuration” on page 167 for more details.
7.
Select whether or not to send a Connected Number IE within the CONNECT message at the
originating ISDN side in the Connected Line Identification Presentation drop-down menu.
The Connected Line Identification Presentation (COLP) is an optional service offered at the
originating interface by the NT to the TE.
Table 151: COLP Parameters
Parameter
8.
Description
Enable
Sends a Connected Number IE within the CONNECT message, which contains the
connected number digits once the transformation of the routing table has been
applied.
Disable
The Connected Number IE is never sent.
Select whether or not to set the Connected Number Information Element restriction at the
destination ISDN side in the Connected Line Identification Restriction drop-down menu.
The Connected Line Identification Restriction (COLR) is a service offered to the TE at the
destination interface.
Table 152: COLR Parameters
Parameter
Dgw v2.0 Application
Description
Enable
When activated at the User UNI-side (Endpoint Type drop-down menu set to TE),
marks the Connected Number IE with a 'restricted' Presentation Indicator, which
keeps privacy over the connected number digits. This option has no effect when
activated at the Network UNI-side (Endpoint Type drop-down menu set to NT).
Disable
No restriction is applied.
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Chapter 23 - ISDN Configuration
9.
Services Configuration
Select whether or not to set the Connected Number Information Element restriction override at the
originating ISDN side in the Connected Line Identification Restriction Override drop-down menu.
Table 153: COLR Override Parameters
Parameter
10.
Description
Enable
When activated at the Network UNI-side (Endpoint Type drop-down menu set to
NT), the connected number digits are delivered even if the Presentation Indicator is
set to 'restricted'. This option has no effect when activated at the User UNI-side
(Endpoint Type drop-down menu set to TE). This is a national option designed for
emergency services.
Disable
No restriction override is applied.
Define whether or not NOTIFY messages can be sent in the Outgoing Notify drop-down menu.
Table 154: Outgoing Notify Parameters
Parameter
Description
Enable
NOTIFY messages can be sent.
Disable
NOTIFY messages are never sent.
The following NOTIFY messages are supported:
11.
•
REMOTE HOLD: Sent when the remote peer holds the call.
•
REMOTE RETRIEVAL: Sent when the remote peer retrieves the call.
Set the Maintenance Service Call Termination drop-down menu with the call termination strategy
after reception of a service message requesting a maintenance on the associated bearer channel.
Table 155: Maintenance Service Parameters
Parameter
12.
Description
Graceful
The call proceeds normally until the user clears the call. The associated bearer is
then set to maintenance. This is the default value.
Abrupt
The call is terminated immediately and set to maintenance.
Define whether or not the optional Date/Time Information Element (IE) can be included in the
CONNECT and SETUP messages in the BRI and PRI.
Table 156: Date/Time IE Support Parameters
Parameter
Description
Disable
Date/Time is not sent
Local Time
Date/Time IE is sent, containing the local time according to the configured time zone in the Network
Host Time Configuration.
UTC
Date/Time IE is sent, containing the Coordinated Universal Time (UTC)
Note: Without a SNTP Synchronized connection, the Date/Time IE is not sent. See “SNTP Configuration”
on page 57 for more details on how to configure the SNTP.
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Software Configuration Guide
13.
Define the AOC-E Support drop-down menu how to send the total charge at the (E)nd of the call in
AOC-E messages.
Table 157: AOC-E Support Parameters
Parameter
no
Description
The AOC-E support is disabled. No information is forwarded to the peer interface.
transparent On an NT interface, the information is sent as received from the network. No
information is sent if the network does not provide information.
On a TE interface, the information is forwarded to the peer interface if AOC
messages are received from the network.
automatic
On an NT interface, always send the information. If the network does not provide
information, 'noChargeAvailable' is sent.
On a TE interface, the information is forwarded to the peer interface if AOC
messages are received from the network.
explicit
On an NT interface, always send the information if the phone requests AOC on a
per-call basis. 'noChargeAvailable' is sent if the network does not provide
information. If the phone does not request AOC on a per-call basis, no information
is sent.
On a TE interface, send an AOC request to the network. If the network rejects the
request, no information is forwarded to the peer interface. Otherwise, the
information is forwarded to the peer interface if AOC messages are received from
the network.
14.
Define the AOC-D Support drop-down menu how to send the current charge (D)uring the call in
AOC-D messages.
Table 158: AOC-D Support Parameters
Parameter
no
Description
The AOC-D support is disabled. No information is forwarded to the peer interface.
transparent On an NT interface, the information is sent as received from the network. No
information is sent if the network does not provide information.
On a TE interface, the information is forwarded to the peer interface if AOC
messages are received from the network.
automatic
On an NT interface, always send the information. If the network does not provide
information, 'noChargeAvailable' is sent.
On a TE interface, the information is forwarded to the peer interface if AOC
messages are received from the network.
explicit
On an NT interface, always send the information if the phone requests AOC on a
per-call basis. 'noChargeAvailable' is sent if the network does not provide
information. If the phone does not request AOC on a per-call basis, no information
is sent.
On a TE interface, send an AOC request to the network. If the network rejects the
request, no information is forwarded to the peer interface. Otherwise, the
information is forwarded to the peer interface if AOC messages are received from
the network.
Note: The AOC features are not available in the NI2 and QSIG signalling protocols. See “PRI Configuration”
on page 155 for more details on how to configure the signalling protocol.
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Chapter 23 - ISDN Configuration
Services Configuration
Note: To enable AOC support on the ISDN interface, you must enable the FACILITY services and at least
one of the following AOC support: AOC-E (End of Call) or AOC-D (During the Call).
Since the AOC from ISDN interface to SIP is currently not supported, enabling the AOC on an ISDN interface
configured as TE (user side) is only meaningful when using hairpinning.
15.
Set the Call Rerouting Behavior drop-down menu with how the call rerouting request received from
the private network side is supported.
The Call Rerouting supplementary service allows to reroute an incoming public ISDN call
(originated from the PSTN) within or beyond the private ISDN network (such a PBX) as specified in
the ETS 300 207 01, section 10.5. The Rerouting data are received and relayed through a Facility
message containing a Facility Information Element.
Rerouting requests are received in a Facility IE.
Table 159: Forward Call Rerouting Parameters
Parameter
Description
Unsupported Rerouting requests received are rejected. A reject answer is sent to the private
network.
Relay
Reroute
Rerouting requests are relayed as received to the public network side. If the peer
rejects or does not support the reroute request, the ISDN service may initiate a
new call to process the rerouting request locally.
Process
Locally
Received Rerouting requests are not relayed to the public network side. The
ISDN service attempts to connect to the rerouted address by initiating a new call.
Note: The Call Rerouting feature is not available in the NI2 and QSIG signalling protocols. See “PRI
Configuration” on page 155 for more details on how to configure the signalling protocol.
16.
Enter one or more numbesr in the MSN field.
You can enter a comma-separated list of numbers. The comma-separated list must use the
following syntax:
777, 888, 999, 555, 444.
This enables the Multiple Subscriber Numbers (MSN) supplementary service with these numbers.
A MSN is a telephone number associated with a line.
The MSN supplementary service enables each individual terminal on one access to have one or
more identities.
If the Called E.164 received from a call does not match any MSN numbers, the call is silently
discarded.
This supplementary service applies only on a BRI Interface configured in TE Point to Multi-Point.
See “BRI Configuration” on page 167 for more details.
17.
188
Click Submit if you do not need to set other parameters.
Dgw v2.0 Application
R2 CAS Parameters
Page Left Intentionally Blank
Introduction
C
Software Configuration Guide
H A P T E R
24
R2 CAS Configuration
This chapter describes how to configure the R2 CAS parameters of the Mediatrix unit.
Note: This chapter applies only to the Mediatrix 3621, Mediatrix 3631, and Mediatrix 3632 models.
Introduction
CAS stands for Channel Associated Signaling. With this method of signalling, each traffic channel has a
dedicated signaling channel. In other words, the signalling for a particular traffic circuit is permanently
associated with that circuit. Channel-associated call-control is still widely used today, mostly in South America,
Africa, Australia, and in Europe.
The Mediatrix unit uses the MFC/R2 CAS protocol. This is a compelled sequence multi-frequency code
signaling. MFC/R2 can be used on international as well as national connections.
In MFC/R2 signaling, the equipment units at the exchanges that send and receive digits, and the signaling
between these units, are usually referred to as register and interregister signalling.
The terms forwards and backwards are heavily used in descriptions of MFC/R2. Forwards is the direction from
the calling party to the called party. Backwards is the direction from the called party to the calling party.
You can configure Mediatrix unit parameters for the E1 R2 CAS.
Line Signals for the Digital Version of MFC/R2
The MFC/R2 digital line signals (defined in ITU-T Q.421) are the ABCD bits of CAS in timeslot 16 of an E1.
They represent the states of the line, and are similar to the states of an analog line. In general, only bits A and
B are used. In most systems, bits C and D are set to fixed values and never change. There are some national
variants where bit C or D may be used for metering pulses.
Interregister Signals
The interregister, or interswitch, signals in MFC/R2 signaling (defined in ITU-T Q.441) are encoded as the
presence of 2, and only 2, out of 6 specific tones, spaced at 120 Hz intervals. Two sets of tones are defined –
one for forward signals, and one for backward signals. There are 15 combinations of 2 out of 6 tones, so there
are 10 signals for the digits 0 to 9, and 5 additional signals available for supervisory purposes.
MFC/R2 uses a separate set of frequencies for the forward and backwards directions.
The interregister signals are sent in-band. They may pass transparently through several nodes in the network
between the two terminating switches. The signals are arranged in groups. When a call begins, the calling end
uses group I signals, and the called end uses group A. The called end may tell the calling end to switch to
using group II and group B signals, or to switch back to group A. In some countries, there are also groups III
and C, used for caller number transfer. Groups III and C do not exist in the ITU specifications.
MFC/R2 uses a system called compelled signaling. To ensure the sending end never sends signals too fast,
each signal from the sending end results in an acknowledgement from the receiving end. The sending end is
instructed signal by signal what it should send next – a dialed digit, a digit of caller ID, etc.
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Chapter 24 - R2 CAS Configuration
Selecting the R2 Signaling Protocol
Selecting the R2 Signaling Protocol
You must set the unit to use the R2 signaling protocol. You can do so in the System / Hardware page. The
Hardware page differs depending on the product and model you have.
 To configure the Mediatrix unit hardware:
1.
In the web interface, click the System link, then the Hardware sub-link.
Figure 71: System – Hardware Web Page
2
2.
3
4
In the PRI Cards Configuration section, select the reference of the clock source in the Clock
Reference drop-down menu.
If you want to configure the clock reference of a specific interface, you must set the Clock Mode
drop-down menu to Master. See “R2 Channel Associated Signaling” on page 194 for more details.
Table 160: Clock Reference
Reference
Description
None
The internal clock does not synchronize with any other source.
Other
Card
The internal clock synchronizes with the other R2 interface of the Mediatrix unit.
This interface must be configured in Slave mode (Clock Mode drop-down menu of
the R2 Channel Associated Signaling section) to provide the clock reference to the
other interfaces.
Note: This choice is not available on the Mediatrix 3621 and 3631 models.
3.
Select whether the line uses T1 or E1 in the Line Type drop-down menu.
Currently, R2 works only on the E1 line type.
Note: Before version 1.1r8.76, the Isdn service needed to be restarted when modifying the Line Type. Since
version 1.1r8.76, the Line Type variable has been moved to the Ex1Pri_1 service and now you rather need
to restart the unit instead of restarting the service when the Line Type is modified.
4.
Select the R2 signaling in the Signaling drop down menu.
When changing from R2 to ISDN or ISDN to R2, you must change your routes accordingly. For
instance, if you are in ISDN with a route isdn-Slot2/E1T1, then change to R2, you must change the
route to r2-Slot2/E1T1.
5.
192
Click Submit if you do not need to set other parameters.
Dgw v2.0 Application
R2 Auto-Configuration
Software Configuration Guide
R2 Auto-Configuration
The R2 Auto-configuration feature allows you to detect and to configure all R2 interfaces so that the R2 link
goes up and becomes usable with a minimal user interaction. When launching an auto-configuration process,
it stops automatically when all interfaces have been tested. For each interface, the auto-configuration process
is considered successful when the link becomes up or a failure when all combinations have been tried without
having a link up.
Caution: Launching the auto-configuration may terminate abruptly all ongoing R2 calls.
Note: Auto-configuration on all R2 interfaces may take some time to complete. Some of the current R2
settings might be replaced by new values.
Please note that some parameters cannot be auto configured. For instance, the clock mode is configured
according to the endpoint type, master for NT and slave for TE.
 To launch the auto-configuration process:
1.
In the web interface, click the R2 link, then the Status sub-link.
Figure 72: R2 – Status Configuration Section
2.
Click the Start Sensing button.
The process starts.
Preset
The R2 Preset Configuration section allows you to load a set of preset configuration for your R2 connections.
These preset files are located in the file system's persistent memory. They differ depending on the Mediatrix
unit you are using. Depending on your unit's profile, it may be possible that no preset files are available.
Using preset files is especially useful for units that do not use the default values provided by Media5 (for
instance, T1 instead of E1 for Mediatrix 3000 units). Please note that only script files work. Any other type of
file present in the file system cannot be run here.
You can also export your current R2 configuration in a preset. Please note that these user-defined presets are
not kept in the event of a partial or factory reset.
To see the content of the unit’s file system persistent memory, go to File Manager (“Chapter 53 - File Manager”
on page 543). All installed configuration scripts/images are listed.
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Chapter 24 - R2 CAS Configuration
R2 Channel Associated Signaling
 To load and execute a preset file:
1.
In the R2 Status tab, R2 Preset Configuration section, select one of the available preset files in the
Local Preset drop-down menu.
Figure 73: R2 – Status Configuration Section
1
2.
Click Apply.
The configuration is applied.
 To export the current R2 configuration as a preset:
1.
In the R2 Preset Configuration section, type a name for the preset in the Preset Name field.
Figure 74: R2 – Status Configuration Section
1
2.
Click Save.
The current R2 configuration is exported. Please note that these user-defined presets are not kept
in the event of a partial or factory reset.
When the clock device is not synchronized, the description value of the file is "Automatically
Generated". When synchronized, the description is "Automatically Genereted on Date/Time". See
the File Manager (“Chapter 53 - File Manager” on page 543) for more details on how to see and
manage the files in the unit’s file system.
Partial Reset
When a partial reset is triggered, the user-defined presets are deleted.
R2 Channel Associated Signaling
The R2 Channel Associated Signaling section allows you to define the general parameters related to R2.
194
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R2 Channel Associated Signaling
Software Configuration Guide
 To configure the R2 CAS parameters:
1.
In the web interface, click the R2 link, then the R2 Channel Associated Signaling Config sub-link.
Figure 75: R2 Channel Associated Signaling Web Page
2
3
4
5
6
7
8
9
10
11
12
13
14
15
2.
16
Select to which interface you want to apply the changes in the Select Interface drop-down menu at
the top of the window.
The number of interfaces available vary depending on the Mediatrix unit model you have.
3.
Select the clock mode of the interface in the Clock Mode drop-down menu.
The interface can either generate the clocking for the line or accept the clock from the line.
Table 161: R2 Interface Clock Mode
Mode
Description
Master The interface generates the clock.
Slave
The interface accepts the clock from the line.
The clock source can be selected from the Clock Reference drop-down menu (see “Selecting the
R2 Signaling Protocol” on page 192 for more details). The clock mode could be used, for instance,
to synchronize several units in master clock mode via an E1 line.
4.
Select the port pinout in the Port Pinout drop-down menu.
Table 162: Port Pinout
Mode
5.
Description
Auto
The pinout is set according to the Clock Mode parameter setting (Step 3).
Te
Forces the pinout to TE regardless of the Clock Mode value.
Nt
Forces the pinout to NT regardless of the Clock Mode value.
Set the Monitor Link State drop-down menu with the physical link state of the R2 interface.
Table 163: Interface Link State
Parameter
Enable
Dgw v2.0 Application
Description
The R2 endpoint's operational state is affected by its interface physical link state.
When the link state of the R2 interface is down, the operational state of its
matching endpoint becomes "disable".
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Chapter 24 - R2 CAS Configuration
R2 Channel Associated Signaling
Table 163: Interface Link State (Continued)
Parameter
Disable
Description
The R2 endpoint's operational state is not affected by its interface physical link
state.
Note that if the Monitor Link State parameter is enabled and the Ignore SIP OPTONS on no usable
endpoints parameter is also enabled in the SIP / Interop page, this will influence how the SIP options
are answered. See “SIP Interop” on page 279 for more details.
6.
Select the transmission encoding of bits in the Line Coding drop-down menu.
Table 164: Transmission Encoding
Coding
Description
B8ZS
Bipolar with 8-Zeros Substitution (T1 lines). Currently not available.
HDB3
High-Density Bipolar with 3-zeros (E1 lines).
AMI
Alternate Mark Inversion (E1 and T1 lines).
Make sure that the transmission encoding matches with the remote system. For further information,
see ITU-T Recommendation G.703.
7.
Select the frame format in the Line Framing drop-down menu.
Line Framing is used to synchronize the channels on the frame relay circuit (when a frame starts
and finishes). Without it, the sending and receiving equipment would not be able to synchronize
their frames.
Table 165: Line Framing
Format
Description
SF(D4)
Super frame. Sometimes known as D4 (T1 lines). Currently not available.
ESF
Extended super frame (T1 lines). Currently not available.
CRC4
Cyclic redundancy check 4 (E1 lines).
NO-CRC4 No Cyclic redundancy check 4 (E1 lines).
For further information, see ITU-T Recommendation G.704.
8.
Define the range of active bearer channels in the Channel Range field.
9.
Select the strategy for selecting bearer channels in the Channel Allocation Strategy drop-down
menu.
Table 166: Channel Allocation Strategy
Allocation
196
Description
Ascending
Starting from the lowest-numbered non-busy bearer channel and
going toward the highest-numbered non-busy bearer channel, the
Mediatrix unit selects the first bearer channel available.
Descending
Starting from the highest-numbered non-busy bearer channel and
going toward the lowest-numbered non-busy bearer channel, the
Mediatrix unit selects the first bearer channel available.
Dgw v2.0 Application
R2 Channel Associated Signaling
Software Configuration Guide
Table 166: Channel Allocation Strategy (Continued)
Allocation
RoundRobinAscending
Description
The Mediatrix unit starts from the bearer channel that follows the
bearer channel used for the last call. For instance, if channel #1 was
used in the last call, the unit starts with channel #2. Going toward the
highest-numbered non-busy bearer channel, the unit selects the first
channel available. If the highest channel is unavailable, the search
continues from the lowest-numbered non-busy bearer channel.
RoundRobinDescending The Mediatrix unit starts from the bearer channel that precedes the
bearer channel used for the last call. For instance, if channel #3 was
used in the last call, the unit starts with channel #2. Going toward the
lowest-numbered non-busy bearer channel, the unit selects the first
channel available. If the lowest channel is unavailable, the search
continues from the highest-numbered non-busy bearer channel.
10.
Define the maximum number of active calls on the interface in the Maximum Active Calls field.
This limits the total number of concurrent calls on the interface. Entering 0 indicates no maximum
number of active calls.
11.
Set the Encoding Scheme drop-down menu with the voice encoding scheme in the bearer
capabilities.
This encoding scheme is used when initiating a call on the R2 side. The supported encoding
schemes are G.711 u-Law and G.711 a-Law.
12.
Select the protocol to use for the line signaling in the Line Signaling Protocol drop-down menu.
This signaling must match the connected equipment or network. The Mediatrix unit currently
supports only the Q421-2BitsSignaling signaling, which is the R2 line signaling type ITU-U Q.421.
It is typically used for PCM systems.
13.
Select the R2 incoming digit signaling method in the Incoming Digit Signaling drop-down menu.
Digit signaling is also known as Address Signaling, selection signals and register signaling. The
digits are used primarily to indicate the called number, but can also have other meanings.
Table 167: Incoming Digit Signaling Parameters
Allocation
14.
Description
MfcR2
Multi Frequency Compelled - R2.
DtmfR2
Dual Tone Multi Frequency - R2.
Select the R2 outgoing digit signaling method in the Outgoing Digit Signaling drop-down menu.
Digit signaling is also known as Address Signaling, selection signals and register signaling. The
digits are used primarily to indicate the called number, but can also have other meanings.
Table 168: Outgoing Digit Signaling Parameters
Allocation
15.
Description
MfcR2
Multi Frequency Compelled - R2.
DtmfR2
Dual Tone Multi Frequency - R2.
Select the country in the Country Selection drop-down menu.
You have the following choices:
Dgw v2.0 Application
•
BrazilR2
•
MexicoR2
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Chapter 24 - R2 CAS Configuration
16.
R2 Signaling Variants
•
ArgentinaR2
•
SaudiArabiaR2
•
VenezuelaR2
•
PhilipinesR2
•
ITU-TR2
Set the Digit Attenuation field with the additional attenuation, in dB, for MFR2/DTMF digits
generation.
By default, MFR2/DTMF digits generation power is determined by country selection. This parameter
provides a mean to reduce this power.
R2 Signaling Variants
This section allows you to decide whether or not you want to override the default R2 signaling parameters.
The Mediatrix unit uses the following default values:
Table 169: R2 Signaling Parameters Default Values
Default Value (ms)
Parameter
Bra.
198
Mex.
Arg.
Sau.
Ven.
Phi.
ITU-T
Bits CD
1
1
1
1
1
1
1
ANI Length
0
(Variable
ANI
length)
0
(Variable
ANI
length)
0
(Variable
ANI
length)
0
(Variable
ANI
length)
0
(Variable
ANI
length)
0
(Variable
ANI
length)
0
(Variable
ANI
length)
DNIS Length
0
(Variable
DNIS
length)
0
(Variable
DNIS
length)
0
(Variable
DNIS
length)
0
(Variable
DNIS
length)
0
(Variable
DNIS
length)
0
(Variable
DNIS
length)
0
(Variable
DNIS
length)
ANI Request
Enable
Disable
Enable
Enable
Enable
Enable
Enable
Send ANI request
after nth DNIS Digits
0
(Variable
number
of DNIS
digits)
0
(Variable
number
of DNIS
digits)
0
(Variable
number
of DNIS
digits)
1
0
(Variable
number
of DNIS
digits)
0
(Variable
number
of DNIS
digits)
0
(Variable
number
of DNIS
digits)
Collect Call Blocked
Enabled
Enable
Disable
Disable
Disable
Disable
Disable
Disable
ANI Category
NatSubsc Nat
riberNoPr Subscrib
er NoPrio
io
Nat
Subscrib
er NoPrio
Nat
Subscrib
er NoPrio
Nat
Subscrib
er NoPrio
Nat
Subscrib
er NoPrio
Nat
Subscrib
er NoPrio
Line Free Category
LineFree
NoCharg
e
Line Free
NoCharg
e
Line Free
Charge
Line Free
Charge
Line Free
Charge
Line Free
Charge
Line Free
Charge
ANI Restricted
Enable
Disable
Disable
Disable
Disable
Disable
Disable
Incoming Decline
Method
Release
Release
Release
Release
Release
Release
Release
Dgw v2.0 Application
R2 Signaling Variants
Software Configuration Guide
Override Default Country Settings
You can override the default R2 signaling parameters. In that case, you will have access to the R2 Signaling
Variants section to define the signaling you want.
 To override the R2 signaling default settings:
1.
In the web interface, click the R2 link, then the Signaling sub-link.
Figure 76: R2 Signaling Variants Web Page
2
3
2.
Select to which interface you want to apply the changes in the Select Interface drop-down menu at
the top of the window.
The number of interfaces available vary depending on the Mediatrix unit model you have.
3.
Select whether or not you want to override the default setting of R2 signaling parameters in the
Override Default Country Settings drop-down menu.
Table 170: R2 Signaling Override
Allocation
Description
Disable
The interface uses the default country configuration.
Enable
The interface uses the specific country configuration as defined in the
R2 Signaling Variants section. To retrieve the default configuration
associated with the current country, click the Reset to Default button.
Proceed to “R2 Signaling Variants” on page 199.
Overriding the default settings is considered as advanced configuration. Media5 recommends not
to modify the country variants unless you know exactly what you are doing.
R2 Signaling Variants
This section allows you to define R2 signaling parameters. You can click the Reset to Default button at any
time to revert back to the default R2 signaling.
 To set R2 signaling parameters:
1.
In the R2 Signaling Variants section, set the C and D bits when the device transmits line signals in
the Bits CD field.
The device ignores the C and D bits of received line signals.
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Chapter 24 - R2 CAS Configuration
R2 Signaling Variants
Figure 77: R2 Signaling Variants Section
1
2
3
4
5
6
7
8
9
10
11
2.
Set the ANI Length field with the length of Automatic Number Identification (ANI) to be requested or
sent.
If a variable-length ANI is used, the End of ANI tone set in “R2 Tones Forward Groups” on page 213
is sent to indicate the end of the ANI digits. When fixed-length ANI is used and the available ANI
digits are longer than the requested length, the last n digits are sent. If available ANI digits are
shorter, then the End of ANI tone set in “R2 Tones Forward Groups” on page 213 is sent.
3.
•
0: Variable ANI length.
•
1..20: Specific ANI length.
Set the DNIS Length field with the length of the Dialed Number Identification Service (DNIS)
expected.
DNIS is the called party or the destination number. If a variable length is defined, then the I-15 digit
is used to indicate the end of DNIS.
4.
•
0: Variable DNIS length used.
•
1..20: Specific DNIS length expected.
Define whether or not ANI should be requested in the ANI Request field.
ANI is the calling party number. When ANI is requested, the calling party category followed by the
actual ANI is sent. If this parameter is enabled, the ANI request is sent after the nth DNIS digit
(defined in Step 5) is received.
5.
Set the Send ANI Request after nth DNIS Digits field with the number of DNIS digits to be received
before sending the ANI request (if the ANI Request field is set to Enable).
If a variable number is used, the ANI request is sent after all DNIS digits have been received.
6.
•
0: Variable number of DNIS digits.
•
1..10: Specific number of DNIS digits.
Define whether or not the Collect Call Blocked Option is used in the Collect Call Blocked Enabled
drop-down menu.
Two methods actually exist to do Collect Call Blockage. The first method refers to R2 signaling and
how, through the use of signals, collect calls can be blocked. The second method refers to how,
through the use of double answer, the same end can be achieved.
For an incoming collect call, a signal of Group II-8 is sent forward from the caller to the called party.
The called party implements the collect call blockage (when enabled) by sending backward to the
caller a signal of Group B-7 indicating that the collect calls are not being accepted by the called
party. Consequently, the originator of the call gets a busy tone and the local calling party circuit that
has been used for the call is dropped when the originator puts the phone on hook.
The double answer allows the destination side to reject or accept a collect call (toll). Since the owner
of the collect call is the person being called, the CO recognizes that the call is being dropped just
by the fact that the call was dropped. For regular, non-collect calls, the owner of the call is the
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R2 Signaling Variants
Software Configuration Guide
person calling and not the party being called. So, if the receiver of the call decides to refuse the call,
a double answer is generated within a specified time. If the receiver wants to answer the call, a
double answer is not generated and the receiver is then billed for the incoming call.
Table 171: Collect Call Blockage Parameters
Parameter
7.
Description
Enable
The signal of Group B-7 is sent if a Group II-8 (Collect Call) signal is
received from the caller or a double answer is generated within a
specified time depending on the value defined in the R2 Timer
Variants table (see “R2 Timers Variants” on page 204 for more
details).
Disable
No signal is sent in response to a Group II-8 (Collect Call) signal and/
or no double answer is generated by the called side upon incoming
calls.
Set the ANI Category drop-down menu with the group II forward signal to be sent upon receiving a
calling party category request.
This tone indicates the category of the calling party.
Table 172: ANI Category Parameters
Parameter
8.
Description
NatSubscriberNoPrio
The call is set up from a national subscriber’s line and is non-priority.
NatSubscriberPrio
The call is set up from a national subscriber’s line to which priority
treatment of calls has been granted.
NatMaintenance
The call comes from a national maintenance equipment.
NatSpare
Spare.
NatOperator
The call is set up from a national operator’s position.
NatData
The call will be used for national data transmission.
IntSubscriberNoPrio
The call is set up from an international subscriber’s line and is nonpriority.
IntData
The call will be used for international data transmission.
IntSubscriberPrio
The call is set up from an international subscriber’s line to which
priority treatment of calls has been granted.
IntOperator
The call is set up from an international operator’s position.
CollectCall
The call is set up for Call Collect.
Set the Line Free Category drop-down menu with the group B backward signal to be sent by the
incoming R2 register to indicate line free condition of the destination party.
Table 173: Line Free Category Parameters
Parameter
9.
Dgw v2.0 Application
Description
LineFreeNoCharge
The called party’s line is free but is not to be charged on answer.
LineFreeCharge
The called party’s line is free but is to be charged on answer.
Set the ANI Restricted drop-down menu with the behaviour of the unit following the reception of a
reject request after sending the Send next digit (ANI) request.
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Chapter 24 - R2 CAS Configuration
R2 Timers Variants
The request is generally rejected when the calling party is unable to send its identification.
Table 174: ANI Restricted Parameters
Parameter
10.
Description
Enable
A congestion tone is sent in response to the reject request and the
call MUST be dropped.
Disable
The unit uses the same behaviour as the End of ANI Tone and the
call WILL be completed.
Set the Incoming Decline Method drop down to indicate how to cancel a call attempt from R2 if the
called party rejects the call when in Seizure Acknowledged state (waiting for answer).
Table 175: Incoming Decline Method
Parameter
Description
Release:
B bit is set to 0 ans state is set to Released.
ClearBack
B bit is set to 0 until decline quard expires then B is set to 1
and state is set to Clear-back.
11.
Click Submit if you do not need to set other parameters.
R2 Timers Variants
This section allows you to decide whether or not you want to override the default R2 timers parameters. The
Mediatrix unit uses the following default values:
Table 176: R2 Timers Default Values
Default Value (ms)
Parameter
Bra.
202
Mex.
Arg.
Sau.
Ven.
Phi.
ITU-T
Seizure Ack Timeout
2000
2000
2000
2000
2000
2000
2000
Fault Seizure Ack
Timeout
60000
60000
60000
60000
60000
60000
60000
Double Seizure
Timeout
100
100
100
100
100
100
100
Double Answer
Timeout
1000
1000
1000
1000
1000
1000
1000
Answer Timeout
0
0
0
0
0
0
0
ReAnswerTimeout
1000
1000
1000
1000
1000
1000
1000
Release Guard
Timeout
100
100
100
100
100
100
100
InterCall Guard
Timeout
100
100
100
100
100
100
100
Congestion Tone
Guard Timeout
1000
1000
1000
1000
1000
1000
1000
Unblocking Timeout
100
100
100
100
100
100
100
Address Complete
Timeout
8000
8000
8000
8000
8000
8000
8000
Dgw v2.0 Application
R2 Timers Variants
Software Configuration Guide
Table 176: R2 Timers Default Values (Continued)
Default Value (ms)
Parameter
Bra.
Mex.
Arg.
Sau.
Ven.
Phi.
ITU-T
Wait Answer Timeout
60000
60000
60000
60000
60000
60000
60000
Digit Complete
Timeout
4000
4000
4000
4000
4000
4000
4000
Wait GroupB
Response Complete
Timeout
3000
3000
3000
3000
3000
3000
3000
Wait Immediate
Response Complete
Timeout
1000
1000
1000
1000
1000
1000
1000
Play Tone Guard
Timeout
70
70
70
70
70
70
70
Accept Call Timeout
2000
2000
2000
2000
2000
2000
2000
Clear Forward Guard
Timeout
1500
1500
1500
1500
1500
1500
1500
Clear Backward
Guard Timeout
1500
1500
1500
1500
1500
1500
1500
Fault On Answered
Guard Timeout
250
250
250
250
250
250
250
Fault On Clear
Backward Guard
Timeout
250
250
250
250
250
250
250
Fault On Seize Ack
Guard Timeout
250
250
250
250
250
250
250
Fault On Seize Guard
Timeout
250
250
250
250
250
250
250
Decline Guard
Timeout
1500
1500
1500
1500
1500
1500
1500
Override Default Country Settings
You can override the default R2 timers. In that case, you will have access to the R2 Timers Variants section
to define the timers you want.
 To override the R2 timers default settings:
1.
In the web interface, click the R2 link, then the Timers sub-link.
Figure 78: R2 Timers Variants Web Page
2
3
Dgw v2.0 Application
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Chapter 24 - R2 CAS Configuration
2.
R2 Timers Variants
Select to which interface you want to apply the changes in the Select Interface drop-down menu at
the top of the window.
The number of interfaces available vary depending on the Mediatrix unit model you have.
3.
Select whether or not you want to override the default setting of R2 timers in the Override Default
Country Settings drop-down menu.
Table 177: R2 Timers Override
Allocation
Description
Disable
The interface uses the default country configuration.
Enable
The interface uses the specific country configuration as defined in the
R2 Timers Variants section. To retrieve the default configuration
associated with the current country, click the Reset to Default button.
Proceed to “R2 Timers Variants” on page 204.
Overriding the default settings is considered as advanced configuration. Media5 recommends not
to modify the country variants unless you know exactly what you are doing.
R2 Timers Variants
This section allows you to define R2 timers. You can click the Reset to Default button at any time to revert back
to the default R2 timers.
204
Dgw v2.0 Application
R2 Timers Variants
Software Configuration Guide
 To set R2 timers:
1.
In the R2 Timers Variants section, set the Seizure ACK Timeout field with the maximum time, in
milliseconds (ms), an outgoing R2 register waits for the seizure acknowledgement signal after
sending a seizure signal.
Figure 79: R2 Timers Variants Section
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
2.
Set the Fault Seizure Ack Timeout field with the maximum time, in milliseconds (ms), an incoming
R2 register waits for a seizure acknowledge failure condition to clear.
3.
Set the Double Seizure Timeout field with the minimum time, in milliseconds (ms), an outgoing R2
register waits after a double seizure is recognized before releasing the connection.
4.
Set the Double Answer Timeout field with the maximum time, in milliseconds (ms), an outgoing R2
register waits after receiving a clear-backward signal before releasing the connection.
5.
Set the Answer Timeout field with the maximum time, in milliseconds, the answer signal AB=01 is
applied before the clear backward signal AB=11 is sent.
This variable is generally used to reject collect call (toll) and is only available if the
CallCollectBlocked is enabled.
A value of 0 means that the signal is applied until the call is disconnected. However, if a special
event (flash hook) is detected in the answered state, then the clear backward signal AB=11 is
immediately applied for a period corresponding to the ReAnswer Timeout parameter.
6.
Dgw v2.0 Application
Set the ReAnswer Timeout field with the maximum time, in milliseconds, the clear backward signal
AB=11 is applied before the answer signal AB=01 is reapplied again.
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Chapter 24 - R2 CAS Configuration
R2 Timers Variants
This variable is generally used to reject collect call (toll) and is only available if the Collect Call
Blocked Enabled parameter is enabled (see “R2 Signaling Variants” on page 199 for more details).
The ReAnswerTimeout is only applied if the AnswerTimeout or an event generating the clear
backward signal is triggered.
7.
Set the Release Guard Timeout field with the maximum time, in milliseconds (ms), an incoming R2
register waits before sending an idle line signal when a clear forward line signal is received.
8.
Set the InterCall Guard Timeout field with the maximum time, in milliseconds (ms), an outgoing R2
register waits after receiving an idle line signal before attempting a new seizure of the line.
9.
Set the Congestion Tone Guard Timeout field with the maximum time, in milliseconds (ms), an
incoming R2 register waits after sending a congestion tone before sending a clear forward line
signal and transit to the idle state.
10.
Set the Unblocking Timeout field with the maximum time, in milliseconds (ms), a both-way trunk
waits before assuming an idle state when a blocking condition is removed.
This will prevent a too aggressive seizure of the trunk.
11.
Set the Address Complete Timeout field with the maximum time, in milliseconds (ms), that the caller
waits for the reception of an Address Complete Tone after sending all ANI or DNIS digits.
12.
Set the Wait Answer Timeout field with the maximal time, in milliseconds (ms), that the caller waits
for an Answer signal (ANSW) after receiving a Group B Line Free Signal Tone.
This timer is effective only when the line is free.
13.
Set the Digit Complete Timeout field with the maximal time, in milliseconds (ms), that the caller waits
for a Group I forward tone after sending either a Group A next DNIS digit, next ANI digit, or next
calling category Tone.
14.
Set the Wait GroupB Response Complete Timeout field with the maximal time, in milliseconds (ms),
that the caller waits for the confirmation of the end of the compelled sequence after receiving a
Group B Signal Tone.
The end of the compelled sequence is detected by a transition of the backward tone to off.
15.
Set the Wait Immediate Response Complete Timeout field with the maximal time, in milliseconds
(ms), that the caller waits for the confirmation of the end of the compelled sequence after receiving
a Group B Signal Tone.
The end of the compelled sequence is detected by a transition of the backward tone to off. This timer
is specific to the immediate accept Signal Tone.
16.
Set the Play Tone Guard Timeout field with the maximum time, in milliseconds (ms), an incoming
R2 register waits after receiving the confirmation of the reception of the Group B Signal Tone before
playing one of the calling tones in the caller direction.
17.
Set the Accept Call Timeout field with the time, in milliseconds (ms), that the unit waits to accept a
R2 CAS call.
18.
Set the Clear Forward Guard Timeout field with the maximum time, in milliseconds (ms), an
outgoing R2 register waits after sending a clear forward line signal before transiting to the idle state.
19.
Set the Clear Backward Guard Timeout field with the maximum time, in milliseconds (ms), an
incoming R2 register waits after sending a clear backward line signal before sending the idle line
signal and transit in the idle state.
20.
Set the Fault On Answered Guard Timeout field with the maximum time, in milliseconds (ms), an
outgoing R2 register waits for the fault to clear before sending the clear forward line signal.
In the case bb = 0 is recognized while in the answered state, no immediate action is taken. However,
the clear forward signal is sent if bb = 1 is restored or the answered guard timeout is reached.
21.
206
Set the Fault On Clear Backward Guard Timeout field with the maximum time, in milliseconds (ms),
an outgoing R2 register waits for the fault to clear before sending the clear forward line signal.
Dgw v2.0 Application
R2 Digit Timers Variants
Software Configuration Guide
In the case bb = 0 is recognized while in the clear backward state, no immediate action is taken.
However, the clear forward signal is sent if bb = 1 is restored or the clear backward guard timeout
is reached.
22.
Set the Fault On Seize Ack Guard Timeout field with the maximum time, in milliseconds (ms), an
outgoing R2 register waits for the fault to clear before sending the clear forward line signal.
In the case bb = 0 is recognized while in the seize acknowledge state prior to the answer signal, no
immediate action is taken. However, the clear forward signal is sent if bb = 1 is restored or the seize
acknowledge guard timeout is reached.
23.
Set the Fault On Seize Guard Timeout field with the maximum time, in milliseconds (ms), an
outgoing R2 register waits for the fault to clear before sending the clear forward line signal.
In the case bb = 0 is recognized while in the seize state, no immediate action is taken. However,
when the seize acknowledgement signal is recognized after the seize ack timeout period has
elapsed or the seize guard timeout is reached, the clear forward signal is sent.
24.
Set the Decline Guard Timeout field to determine the maximum time the AB=10 release signal is
applied before sending the AB=11clearback signal. This variable applies when
IncomingDeclineMethod is set to ClearBack while declining a call. The value is expressed in
milliseconds (ms).
25.
Set the NoDigitTimeout field to the Maximum time an incoming R2 register waits before sending the
congestion tone when no digits are received.
26.
This value is expressed in milliseconds (ms
27.
Click Submit if you do not need to set other parameters.
R2 Digit Timers Variants
This section allows you to decide whether or not you want to override the default R2 digit timers. The Mediatrix
unit uses the following default values:
Table 178: R2 Digit Timers Default Values
Default Value (ms)
Parameter
Bra.
Mex.
Arg.
Sau.
Ven.
Phi.
ITU-T
MFC Pulse Inter Digit
Timeout
100
100
100
100
100
100
100
MFC Pulse Min On
Timeout
150
150
150
150
150
150
150
MFC Max Sequence
Timeout
10000
20000
20000
20000
20000
20000
20000
MFC Max On Timeout 5000
10000
10000
10000
10000
10000
10000
MFC Max Off Timeout 5000
10000
10000
10000
10000
10000
10000
Override Default Country Settings
You can override the default R2 digit timers. In that case, you will have access to the R2 Digit Timers Variants
section to define the timers you want.
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Chapter 24 - R2 CAS Configuration
R2 Digit Timers Variants
 To override the R2 digit timers default settings:
1.
In the web interface, click the R2 link, then the Digit Timers sub-link.
Figure 80: R2 Digit Timers Variants Web Page
2
3
2.
Select to which interface you want to apply the changes in the Select Interface drop-down menu at
the top of the window.
The number of interfaces available vary depending on the Mediatrix unit model you have.
3.
Select whether or not you want to override the default setting of R2 digit timers in the Override
Default Country Settings drop-down menu.
Table 179: R2 Digit Timers Override
Allocation
Description
Disable
The interface uses the default country configuration.
Enable
The interface uses the specific country configuration as defined in the
R2 Digit Timers Variants section. To retrieve the default configuration
associated with the current country, click the Reset to Default button.
Proceed to “R2 Digit Timers Variants” on page 208.
Overriding the default settings is considered as advanced configuration. Media5 recommends not
to modify the country variants unless you know exactly what you are doing.
R2 Digit Timers Variants
This section allows you to define R2 digit timers. You can click the Reset to Default button at any time to revert
back to the default R2 digit timers.
 To set R2 digit timers:
1.
In the R2 Digit Timers Variants section, set the MFC Pulse Inter Digit Timeout field with the minimum
delay, in milliseconds (ms), between the end of transmission of the last signal of the compelled cycle
and the start of the next one.
Figure 81: R2 Digit Timers Variants Section
1
2
3
4
5
6
208
2.
Set the MFC Pulse Min On Timeout field with the minimum time, in milliseconds (ms), a backward
tone can be on from the backward perspective.
3.
Set the MFC Max Sequence Timeout field with the maximum time, in milliseconds (ms), for a
complete compelled signaling cycle from the forward perspective.
Dgw v2.0 Application
R2 Link Timers Variants
Software Configuration Guide
4.
Set the MFC Max On Timeout field with the maximum time, in milliseconds (ms), a forward tone can
be on from the forward perspective.
5.
Set the MFC Max Off Timeout field with the maximum time, in milliseconds (ms), a forward tone can
be off from the forward perspective.
6.
Set the MF Congestion Tone Duration field with the maximum time, in milliseconds (ms), the
transmission of the congestion tone can last.
7.
This value is expressed in milliseconds (ms).
8.
Click Submit if you do not need to set other parameters.
R2 Link Timers Variants
This section allows you to decide whether or not you want to override the default R2 link timers. The Mediatrix
unit uses the following default values:
Table 180: R2 Link Timers Default Values
Default Value (ms)
Parameter
Bra.
Mex.
Arg.
Sau.
Ven.
Phi.
ITU-T
Link Activation
Timeout
1000
1000
1000
1000
1000
1000
1000
Link Activation Retry
Timeout
3000
3000
3000
3000
3000
3000
3000
Override Default Country Settings
You can override the default R2 link timers. In that case, you will have access to the R2 Link Timers Variants
section to define the timers you want.
 To override the R2 link timers default settings:
1.
In the web interface, click the R2 link, then the Link Timers sub-link.
Figure 82: R2 Link Timers Variants Web Page
2
3
2.
Select to which interface you want to apply the changes in the Select Interface drop-down menu at
the top of the window.
The number of interfaces available vary depending on the Mediatrix unit model you have.
Dgw v2.0 Application
209
Chapter 24 - R2 CAS Configuration
3.
R2 Tones Variants
Select whether or not you want to override the default setting of R2 link timers parameters in the
Override Default Country Settings drop-down menu.
Table 181: R2 Link Timers Override
Allocation
Description
Disable
The interface uses the default country configuration.
Enable
The interface uses the specific country configuration as defined in the
R2 Link Timers Variants section. To retrieve the default configuration
associated with the current country, click the Reset to Default button.
Proceed to “R2 Link Timers Variants” on page 210.
Overriding the default settings is considered as advanced configuration. Media5 recommends not
to modify the country variants unless you know exactly what you are doing.
R2 Link Timers Variants
This section allows you to define R2 link timers. You can click the Reset to Default button at any time to revert
back to the default R2 link timers.
 To set R2 link timers:
1.
In the R2 Link Timers Variants section, set the Link Activation Timeout field with the maximum time,
in milliseconds (ms), the unit waits for an activation indication coming from the physical link.
The activation indication is used to indicate that the physical layer connection has been activated.
Figure 83: R2 Link Timers Variants Section
1
2
2.
Set the Link Activation Retry Timeout field with the maximum time, in milliseconds (ms), the unit
waits before attempting to re-establish the physical link.
The attempt is made when the physical layer connection has been deactivated.
3.
Click Submit if you do not need to set other parameters.
R2 Tones Variants
This section allows you to decide whether or not you want to override the default R2 tones parameters. The
Mediatrix unit uses the following default values:
Table 182: R2 Tones Default Values
Default Value (ms)
Parameter
Bra.
Mex.
Arg.
Sau.
Ven.
Phi.
ITU-T
R2 Tones Forward Groups
End of DNIS Tone
I15
None
I15
I15
None
I15
I15
End of ANI Tone
I15
I15
I15
I15
I15
I15
I15
Restricted ANI Tone
I12
None
None
None
I12
None
None
R2 Tones Backward Groups
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R2 Tones Variants
Software Configuration Guide
Table 182: R2 Tones Default Values (Continued)
Default Value (ms)
Parameter
Bra.
Dgw v2.0 Application
Mex.
Arg.
Sau.
Ven.
Phi.
ITU-T
Send Next DNIS Digit
Tone
A1
A1
A1
A1
A1
A1
A1
Send Previous DNIS
Digit Tone
A9
None
A9
A2
A2
A9
A9
Switch to Group II
Tone
A3
A3
A3
A3
A3
A3
A3
Network Congestion
Tone
A4
A4
A4
A4
A4
A4
A4
Send Calling Party
Category Tone
A5
None
A5
A5
A5
A5
A5
Immediate Accept
Tone
None
None
A6
A6
A6
A6
A6
Send DNIS Digit N-2
Tone
A7
None
A7
A7
A7
A7
A7
Send DNIS Digit N-3
Tone
A8
None
A8
A8
A8
A8
A8
Repeat all DNIS Tone
None
None
A10
None
A10
A10
A10
Send Next ANI Digit
Tone
A5
A2
A5
A5
A9
A5
A5
Send Calling Party
Category Tone Switch
to Group C
None
A6
None
None
None
None
None
Send Special
Information Tone
None
None
B2
B2
B2
B2
B2
User Busy Tone
B3
B2
B3
B3
B3
B3
B3
Network Congestion
Tone
B4
B4
B4
B4
B4
B4
B4
Unassigned Number
Tone
B8
None
B7
B5
B5
B5
B5
Line Free with Charge
Tone
B1
B1
B6
B1
B6
B6
B6
Line Free with Charge B6
Tone (Supplementary)
None
None
None
None
None
None
Line Free without
Charge Tone
B7
B5
B7
B6
B7
B7
B7
Line Out of Order
Tone
B8
None
B8
B8
B8
B8
B8
Changed Number
Tone
B3
None
None
None
None
None
None
Send Next ANI Digit
Tone
None
C1
None
None
None
None
None
Repeat All DNIS Tone
switch to Group A
None
C2
None
None
None
None
None
211
Chapter 24 - R2 CAS Configuration
R2 Tones Variants
Table 182: R2 Tones Default Values (Continued)
Default Value (ms)
Parameter
Bra.
Mex.
Arg.
Sau.
Ven.
Phi.
ITU-T
Send Next DNIS Digit
Tone switch to Group
A
None
C5
None
None
None
None
None
Network Congestion
Tone
None
C4
None
None
None
None
None
Send Previous DNIS
Digit Tone switch to
Group A
None
C6
None
None
None
None
None
Switch to Group II
Tone
None
C3
None
None
None
None
None
Override Default Country Settings
You can override the default R2 tones. In that case, you will have access to the R2 Tones Forward Groups
and R2 Tones Backward Groups sections to define the tone you want.
 To override the R2 tones default settings:
1.
In the web interface, click the R2 link, then the Tones sub-link.
Figure 84: R2 Tones Variants Web Page
2
3
2.
Select to which interface you want to apply the changes in the Select Interface drop-down menu at
the top of the window.
The number of interfaces available vary depending on the Mediatrix unit model you have.
3.
Select whether or not you want to override the default setting of R2 tones parameters in the Override
Default Country Settings drop-down menu.
Table 183: R2 Tones Override
Allocation
Description
Disable
The interface uses the default country configuration.
Enable
The interface uses the specific country configuration as defined in the
R2 Tones Forward Groups and R2 Tones Backward Groups sections.
To retrieve the default configuration associated with the current
country, click the Reset to Default button. Proceed to “R2 Tones
Forward Groups” on page 213.
Overriding the default settings is considered as advanced configuration. Media5 recommends not
to modify the country variants unless you know exactly what you are doing.
212
Dgw v2.0 Application
R2 Tones Variants
Software Configuration Guide
R2 Tones Forward Groups
These tones fall into Groups I and II. All Group I and II tones use the ITU-T R2 forward group tone frequencies.
Most of the tones in Group I are used for addressing and identification. Group II tones give information about
the origin of the call. Group II tones are also sent by an outgoing MFC/R2- or International MFC/R2-register in
response to one of the following backward signals:


Change over to the reception of Group B signals.
Send calling party's category.
Table 184: R2 Group I Forward Tones
MF
Tone
Description
1
I-1
Digit 1 (Language: French, if first signal sent in intl. link)
2
I-2
Digit 2 (Language: English, if first signal sent intl. link)
3
I-3
Digit 3 (Language: German, if first signal sent in intl. link)
4
I-4
Digit 4 (Language: Russian, if first signal sent in intl. link)
5
I-5
Digit 5 (Language: Spanish, if first signal sent in intl. link)
6
I-6
Digit 6 (Language: Spare, if first signal sent in intl. link)
7
I-7
Digit 7 (Language: Spare, if first signal sent in intl. link)
8
I-8
Digit 8 (Language: Spare, if first signal sent in intl. link)
9
I-9
Digit 9 (Discriminating digit, if first signal sent in intl. link)
10
I-10
Digit 0 (Discriminating digit, if first signal sent in intl. link)
11
I-11
Country code indicator, outgoing half-echo suppressor required
12
I-12
Country code indicator, no echo suppressor required
13
I-13
Test call indicator (call by automatic test equipment)
14
I-14
Country code indicator, outgoing half-echo suppressor inserted
15
I-15
Signal is not used
You can click the Reset to Default button at any time to revert back to the default R2 tones.
 To set the R2 tones forward groups:
1.
In the R2 Tones Forward Groups section, select the forward Group 1 tone used to send after the
DNIS digits in the End of DNIS Tone drop-down menu.
You have the choice between none and I1 to I15 Group 1 forward signals MF Tone.
Figure 85: R2 Tones – Forward Groups Section
1
3
2.
2
Select the forward Group 1 tone used to send after the ANI digits in the End of ANI Tone drop-down
menu.
You have the choice between none and I1 to I15 Group 1 forward signals MF Tone.
3.
Select the forward Group 1 tone used to reject a query in the Restricted ANI Tone drop-down menu.
This tone is generally used in response to the identification request when the caller party is unable
to send his identification to the called party. If no tone is defined, the End of ANI tone is sent to the
caller.
You have the choice between none and I1 to I15 Group 1 forward signals MF Tone.
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Chapter 24 - R2 CAS Configuration
4.
R2 Tones Variants
Click Submit if you do not need to set other parameters.
R2 Tones Backward Groups
Backward tones fall into Groups A and B. The Group A tones acknowledge Group I forward signals. Under
certain conditions, Group A tones also acknowledge Group II tones. Group B tones convey the following
information to an outgoing MFC/R2 register:


The condition of the switch equipment in the incoming exchange.
The condition of the called subscriber's line.
Table 185: R2 Group A Backward Tones
MF
Tone
Description
1
A-1
Send next digit (n + 1)
2
A-2
Send last but one digit (n -1)
3
A-3
Address-complete, changeover to reception of Group B signals
4
A-4
Congestion in the national network
5
A-5
Send calling party’s category
6
A-6
Address-complete, charge, set-up speech conditions
7
A-7
Send last but two digit (n - 2)
8
A-8
Send last but three digit (n - 3)
9
A-9
Spare for national use
10
A-10
Spare for national use
11
A-11
Send country code indicator
12
A-12
Send language or discrimination digit
13
A-13
Send nature of circuit
14
A-14
Request for information on use of an echo suppressor
15
A-15
Congestion in an international exchange or at its output
Table 186: R2 Group B Backward Tones
MGF Tone
214
Description
1
B-1
Spare for national use
2
B-2
Send special information tone
3
B-3
Subscriber’s line busy
4
B-4
Congestion
5
B-5
Unallocated number
6
B-6
Subscriber’s line free, charge
7
B-7
Subscriber’s line free, no charge
8
B-8
Subscriber’s line out of order
9
B-9
Spare for national use
10
B-10
Spare for national use
11
B-11
Spare for national use
12
B-12
Spare for national use
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Software Configuration Guide
Table 186: R2 Group B Backward Tones (Continued)
MGF Tone
Description
13
B-13
Spare for national use
14
B-14
Spare for national use
15
B-15
Spare for national use
Table 187: R2 Group C Backward Tones
MGF Tone
Description
1
C-1
Send next number of Subscriber A
2
C-2
Send first digit BNUM
3
C-3
Send Group II and Group B signals
4
C-4
Congestion
5
C-5
Send next digit BNUM
6
C-6
Repeat last digit BNUM
7
C-7
Spare for national use
8
C-8
Spare for national use
9
C-9
Spare for national use
10
C-10
Spare for national use
11
C-11
Spare for national use
12
C-12
Spare for national use
13
C-13
Spare for national use
14
C-14
Spare for national use
15
C-15
Spare for national use
 To set the R2 tones backward groups:
1.
In the R2 Tones Backward Groups section, select the backward Group A tone used to request the
next DNIS digit in the Send Next DNIS Digit Tone drop-down menu.
You have the choice between none and A1 to A15 Group A backward signals MF Tone.
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Chapter 24 - R2 CAS Configuration
R2 Tones Variants
Figure 86: R2 Tones – Backward Groups Section
1
3
5
2
4
6
7
9
11
13
15
8
10
12
14
16
17
18
19
20
21
22
23
24
25
26
2.
Select the backward Group A tone used to request the previous DNIS digit in the Send Previous
DNIS Digit Tone drop-down menu.
You have the choice between none and A1 to A15 Group A backward signals MF Tone.
3.
Select the backward Group A tone used to request to the caller a switch of Group II signals in the
Switch to Group II Tone drop-down menu.
You have the choice between none and A1 to A15 Group A backward signals MF Tone.
4.
Select the backward Group A tone to be sent when a congestion network is detected in the Network
Congestion Tone drop-down menu.
You have the choice between none and A1 to A15 Group A backward signals MF Tone.
5.
Select the backward Group A tone sent when the backward group requests the calling party
category in the Send Calling Party Category Tone drop-down menu.
You have the choice between none and A1 to A15 Group A backward signals MF Tone.
6.
Select the backward Group A tone sent when the backward group accepts the call immediately in
the Immediate Accept Tone drop-down menu.
You have the choice between none and A1 to A15 Group A backward signals MF Tone.
7.
Select the backward Group A tone used to request the previous - 1 DNIS digit in the Send DNIS
Digit N-2 Tone drop-down menu.
You have the choice between none and A1 to A15 Group A backward signals MF Tone.
8.
Select the backward Group A tone used to request the previous - 2 DNIS digit in the Send DNIS
Digit N-3 Tone drop-down menu.
You have the choice between none and A1 to A15 Group A backward signals MF Tone.
9.
Select the backward Group A tone used to request all DNIS digits in the Repeat All DNIS Tone dropdown menu.
You have the choice between none and A1 to A15 Group A backward signals MF Tone.
10.
Select the backward Group A tone used to request the next ANI digit in the Send Next ANI Digit
Tone drop-down menu.
You have the choice between none and A1 to A15 Group A backward signals MF Tone.
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Software Configuration Guide
11.
Select the backward Group A tone used to request the calling party category and then switch to
Group C signals in the Send Calling Party Category Tone Switch to Group C drop-down menu.
You have the choice between none and A1 to A15 Group A backward signals MF Tone.
12.
Select the backward Group B tone used to send a special information tone in the Send Special
Information Tone drop-down menu.
You have the choice between none and B1 to B15 Group B backward signals MF Tone.
13.
Select the backward Group B tone used to signal a user busy in the User Busy Tone drop-down
menu.
You have the choice between none and B1 to B15 Group B backward signals MF Tone.
14.
Select the backward Group B tone used to signal a network congestion in the Network Congestion
Tone drop-down menu.
You have the choice between none and B1 to B15 Group B backward signals MF Tone.
15.
Select the backward Group B tone used to signal a unassigned number in the Unassigned Number
Tone drop-down menu.
You have the choice between none and B1 to B15 Group B backward signals MF Tone.
16.
Select the backward Group B tone used to signal that the line is free and charge must be applied in
the Line Free with Charge Tone drop-down menu.
You have the choice between none and B1 to B15 Group B backward signals MF Tone.
17.
Select a supplementary backward Group B tone used to detect that the line is free and charge must
be applied in the Line Free with Charge Tone (Supplementary) drop-down menu.
This is a supplementary (optional) tone used on the reception only. You have the choice between
none and B1 to B15 Group B backward signals MF Tone.
18.
Select the backward Group B tone used to signal that the line is free and no charges must be
applied in the Line Free without Charge Tone drop-down menu.
You have the choice between none and B1 to B15 Group B backward signals MF Tone.
19.
Select the backward Group B tone used to signal that the line is out of order in the Line Out of Order
Tone drop-down menu.
You have the choice between none and B1 to B15 Group B backward signals MF Tone.
20.
Select the backward Group B tone used to signal that the subscriber has changed number in the
Changed Number Tone field.
You have the choice between none and B1 to B15 Group B backward signals MF Tone.
21.
Select the backward Group C tone used to request the next ANI digit in the Send Next ANI Digit
Tone drop-down menu.
You have the choice between none and C1 to C15 Group C backward signals MF Tone.
22.
Select the backward Group C tone used to request all DNIS digits and then switch to Group A
signals in the Repeat All DNIS Tone switch to Group A drop-down menu.
You have the choice between none and C1 to C15 Group C backward signals MF Tone.
23.
Select the backward Group C tone used to request the next DNIS digit and then switch to Group A
signals in the Send Next DNIS Digit Tone switch to Group A drop-down menu.
You have the choice between none and C1 to C15 Group C backward signals MF Tone.
24.
Select the backward Group C tone used to signal a network congestion in the Network Congestion
Tone drop-down menu.
You have the choice between none and C1 to C15 Group C backward signals MF Tone.
25.
Dgw v2.0 Application
Select the backward Group C tone used to request the previous DNIS digit and then switch to Group
A signals in the Send Previous DNIS Digit Tone switch to Group A drop-down menu.
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Chapter 24 - R2 CAS Configuration
PRI R2 CAS Statistics
You have the choice between none and C1 to C15 Group C backward signals MF Tone.
26.
Select the backward Group A tone used to request to the caller a switch of Group II signals in the
Switch to Group II Tone drop-down menu.
You have the choice between none and C1 to C15 Group C backward signals MF Tone.
27.
Click Submit if you do not need to set other parameters.
PRI R2 CAS Statistics
This section describes configuration that is available only in the MIB parameters of the Mediatrix unit. You can
configure these parameters as follows:



by using a MIB browser
by using the CLI
by creating a configuration script containing the configuration variables
The Mediatrix unit collects meaningful statistics for each PRI digital card that can be read via SNMP and CLI.
The Mediatrix unit collects statistics for each of its two cards, if available. Slot 2 and Slot 3 indicate the physical
location of the cards in the unit, Slot 2 being on the left when looking at the rear of the unit.
 To view the PRI statistics:
1.
In the ex1Pri_1MIB, locate the statisticsGroup and expand it.
The following table describes the statistics available.
Table 188: R2 CAS Statistics Displayed
Statistic
statsInfoFramesTra
nsmitted
Description
Number of HDLC frames transmitted.
Note: The term frames does not refer to the structure defined in I.431.
statsInfoFramesRec Number of HDLC frames received.
eived
Note: The term frames does not refer to the structure defined in I.431.
statsInfoOctetsTran
smitted
Number of octets transmitted. This value is obtained by cumulating the
octets transmitted in the HDLC frames.
Note: The term frames does not refer to the structure defined in I.431.
statsInfoOctetsRece Number of octets received. This value is obtained by cumulating the
ived
octets received in the HDLC frames.
Note: The term frames does not refer to the structure defined in I.431.
statsInfoFCSErrors
Frame check sequence (FCS) errors indicate that frames of data are
being corrupted during transmission. FCS error count is the number of
frames that were received with a bad checksum (CRC value) in the HDLC
frame. These frames are dropped and not propagated in the upper layers.
This value is available on E1 and T1.
statsInfoFramesDro
pped
Number of frames dropped. This value is obtained by cumulating the
number of frames dropped when transferring the data from the framer
chip to the device internal buffer.
This value is available on E1 and T1.
statsInfoOctetsDrop
ped
Number of octets dropped. This value is obtained by cumulating the
number of octets dropped when transferring the data from the framer chip
to the device internal buffer.
This value is available on E1 and T1.
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Software Configuration Guide
Table 188: R2 CAS Statistics Displayed (Continued)
Statistic
statsInfoNegativeFr
ameSlipsTransmitte
d
Description
A frame is skipped when the frequency of the transmit clock is greater
than the frequency of the transmit system interface working clock based
on 2.048 MHz (on E1) or 1.544 MHz (on T1).
This value is available on E1 and T1.
statsInfoNegativeFr
ameSlipsReceived
A frame is skipped when the frequency of the received route clock is
greater than the frequency of the receive system interface working clock
based on 2.048 MHz (on E1) or 1.544 MHz (on T1).
This value is available on E1 and T1.
statsInfoPositiveFra
meSlipsTransmitted
A frame is repeated when the frequency of the transmit clock is less than
the frequency of the transmit system interface working clock based on
2.048 MHz (on E1) or 1.544 MHz (on T1).
This value is available on E1 and T1.
statsInfoPositiveFra
meSlipsReceived
A frame is repeated when the frequency of the receive route clock is less
than the frequency of the receive system interface working clock based
on 2.048 MHz (on E1) or 1.544 MHz (on T1).
This value is available on E1 and T1.
statsInfoFramingErr
or
The framing error count indicates that a FAS (Frame Alignment Signal)
word has been received with an error.
The FAS-bits are present in every even frame of timeslot 0 on E1.
The FAS-bits are present in ESF format on T1.
This value is available on E1 and T1.
statsInfoCodeViolati
ons
The code violations count indicates that an encoding error on the PCM
line has been detected.
This value is available on E1 and T1.
statsInfoCRCErrors
The CRC errors count is incremented when a multiframe has been
received with a CRC error.
The CRC error count is available in CRC multiframe mode only on E1.
The CRC error count is in ESF format on T1.
statsInfoE-BitError
The E-Bit error count gives information about the outgoing transmit PCM
line if the E-bits are used by the remote end for submultiframe error
indication. Incrementing is only possible in the multiframe synchronous
state.
Due to signaling requirements, the E-bits of frame 13 and frame 15 of the
CRC multiframe can be used to indicate an error in a received
submultiframes:
Submultiframe I status E-bit located in frame 13
Submultiframe II status E-bit located in frame 15
no CRC error : E = 1
CRC error : E = 0
This value is only available in E1.
statsInfoBlockError
The Block Error count is incremented once per multiframe if a multirame
has been received with a CRC error or a bad frame alignment has been
detected.
This value is only available for ESF format on T1 only.
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PRI R2 CAS Statistics
 To reset the statistics:
1.
In the ex1Pri_1MIB, set the statsInfoResetStats variable to 10: resetStats.
You can also use the following line in the CLI or a configuration script:
ex1Pri_1.statsInfoResetStats=10
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E&M CAS Parameters
Page Left Intentionally Blank
Introduction
C
Software Configuration Guide
H A P T E R
25
E&M CAS Configuration
This chapter describes how to configure the E&M CAS parameters of the Mediatrix unit.
Note: This chapter applies to the following models:
• Mediatrix 3531
• Mediatrix 3532
• Mediatrix 3621
• Mediatrix 3631
• Mediatrix 3632
Introduction
CAS stands for Channel Associated Signaling. With this method of signaling, each traffic channel has a
dedicated signaling channel. In other words, the signaling for a particular traffic circuit is permanently
associated with that circuit. Channel-associated call-control is still widely used today, mostly in South America,
Africa, Australia, and in Europe.
E&M (earth & magneto, or ear & mouth) is a type of CAS signalling that defines line signaling and register
signaling. It is also called Signalling System R1 and is mainly used in North America.
E&M was originally developed to allow PABXs in different geographic locations to communicate over an
analog private circuit. Some digital interfaces such as CAS also use versions of E&M signaling.
The terms forwards and backwards are heavily used in descriptions of E&M. Forwards is the direction from
the calling party to the called party. Backwards is the direction from the called party to the calling party.
You can configure Mediatrix unit parameters for the E1/T1 E&M CAS.
Line Signals for the Digital Version of E&M
Line signalling uses the ABCD bits of CAS. Several types of E&M signalling exist.
Line Signals for the Analogue Version of E&M
E&M has its roots in older analogue signalling. The digital versions of E&M replicate the operation of older
analogue signalling versions.
Inter-Register Signals (Defined in ITU-T Q.310-Q.332)
Inter-register signalling uses either R1 tones (defined in Q.310-Q.332) or DTMF depending on the signalling
type and detailed user-defined variant configuration.
The inter-register signals are sent in-band. They may pass transparently through several nodes in the network
between the two terminating switches.
Selecting the E&M Signalling Protocol
You must set the unit to use the E&M signalling protocol. You can do so in the System / Hardware page. The
Hardware page differs depending on the product and model you have.
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Chapter 25 - E&M CAS Configuration
E&M Auto-Configuration
 To configure the Mediatrix unit hardware:
1.
In the web interface, click the System link, then the Hardware sub-link.
Figure 87: System – Hardware Web Page
2
2.
3
4
In the PRI Cards Configuration section, select the reference of the clock source in the Clock
Reference drop-down menu.
If you want to configure the clock reference of a specific interface, you must set the Clock Mode
drop-down menu to Master. See “E&M Channel Associated Signaling” on page 226 for more
details.
Table 189: Clock Reference
Reference
Description
None
The internal clock does not synchronize with any other source.
Other
Card
The internal clock synchronizes with the other E&M interface of the Mediatrix unit.
This interface must be configured in Slave mode (Clock Mode drop-down menu of
the E&M Channel Associated Signaling section) to provide the clock reference to
the other interfaces.
Note: This choice is not available on the Mediatrix 3521, 3621 and 3631 models.
3.
Select whether the line uses T1 or E1 in the Line Type drop-down menu.
4.
Select the E&M signaling in the Signaling drop down menu.
When changing from E&M to ISDN/R2 or ISDN/R2 to E&M, you must change your routes
accordingly. For instance, if you are in ISDN with a route isdn-Slot2/E1T1, then change to E&M, you
must change the route to e&m-Slot2/E1T1.
5.
Click Submit if you do not need to set other parameters.
E&M Auto-Configuration
The E&M Auto-configuration feature allows you to detect and to configure all E&M interfaces so that the E&M
link goes up and becomes usable with a minimal user interaction. When launching an auto-configuration
process, it stops automatically when all interfaces have been tested. For each interface, the autoconfiguration process is considered successful when the link becomes up or a failure when all combinations
have been tried without having a link up.
Caution: Launching the auto-configuration may terminate abruptly all ongoing E&M calls.
Note: Auto-configuration on all E&M interfaces may take some time to complete. Some of the current E&M
settings might be replaced by new values.
224
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Preset
Software Configuration Guide
Please note that some parameters cannot be auto configured. For instance, the clock mode is configured
according to the endpoint type, master for NT and slave for TE.
 To launch the auto-configuration process:
1.
In the web interface, click the E&M link, then the Status sub-link.
Figure 88: E&M – Status Configuration Section
2.
Click the Start Sensing button.
The process starts.
Preset
The E&M Preset Configuration section allows you to load a set of preset configuration for your E&M
connections. These preset files are located in the file system's persistent memory. They differ depending on
the Mediatrix unit you are using. Depending on your unit's profile, it may be possible that no preset files are
available.
Using preset files is especially useful for units that do not use the default values provided by Media5 (for
instance, T1 instead of E1 for Mediatrix 3000 units). Please note that only script files work. Any other type of
file present in the file system cannot be run here.
You can also export your current E&M configuration in a preset. Please note that these user-defined presets
are not kept in the event of a partial or factory reset.
To see the content of the unit’s file system persistent memory, go to File Manager (“Chapter 53 - File Manager”
on page 543). All installed configuration scripts/images are listed.
 To load and execute a preset file:
1.
In the E&M Status tab, E&M Preset Configuration section, select one of the available preset files in
the Local Preset drop-down menu.
Figure 89: E&M – Status Configuration Section
1
2.
Click Apply.
The configuration is applied.
 To export the current E&M configuration as a preset:
1.
In the E&M Preset Configuration section, type a name for the preset in the Preset Name field.
Figure 90: E&M – Status Configuration Section
1
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Chapter 25 - E&M CAS Configuration
2.
E&M Channel Associated Signaling
Click Save.
The current E&M configuration is exported. Please note that these user-defined presets are not kept
in the event of a partial or factory reset.
When the clock device is not synchronized, the description value of the file is "Automatically
Generated". When synchronized, the description is "Automatically Genereted on Date/Time". See
the File Manager (“Chapter 53 - File Manager” on page 543) for more details on how to see and
manage the files in the unit’s file system.
Partial Reset
When a partial reset is triggered, the user-defined presets are deleted.
E&M Channel Associated Signaling
The E&M Channel Associated Signaling section allows you to define the general parameters related to E&M.
 To configure the E&M CAS parameters:
1.
In the web interface, click the E&M link, then the E&M Channel Associated Signaling Config sublink.
Figure 91: E&M Channel Associated Signaling Web Page
2
3
4
5
6
7
9
8
10
11
13
2.
12
Select to which interface you want to apply the changes in the Select Interface drop-down menu at
the top of the window.
The number of interfaces available vary depending on the Mediatrix unit model you have.
3.
Select the clock mode of the interface in the Clock Mode drop-down menu.
The interface can either generate the clocking for the line or accept the clock from the line.
Table 190: E&M Interface Clock Mode
Mode
Description
Master The interface generates the clock.
Slave
The interface accepts the clock from the line.
The clock source can be selected from the Clock Reference drop-down menu (see “Selecting the
E&M Signalling Protocol” on page 223 for more details). The clock mode could be used, for
instance, to synchronize several units in master clock mode via an E1 line.
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4.
Software Configuration Guide
Select the port pinout in the Port Pinout drop-down menu.
Table 191: Port Pinout
Mode
Auto
Description
The pinout is set according to the Clock Mode parameter setting (Step 3).
Master Forces the pinout to Master regardless of the clock mode.
Slave
5.
Forces the pinout to Slave regardless of the clock mode.
Set the Monitor Link State drop-down menu with the physical link state of the E&M interface.
Table 192: Interface Link State
Parameter
Description
Enable
The E&M endpoint's operational state is affected by its interface physical link state.
When the link state of an E&M interface is down, the operational state of its
matching endpoint becomes "disable".
Disable
The E&M endpoint's operational state is not affected by its interface physical link
state
Note that if the Monitor Link State parameter is enabled and the Ignore SIP OPTONS on no usable
endpoints parameter is also enabled in the SIP / Interop page, this will influence how the SIP options
are answered. See “SIP Interop” on page 279 for more details.
6.
Select the transmission encoding of bits in the Line Coding drop-down menu.
Table 193: Transmission Encoding
Coding
Description
B8ZS
Bipolar with 8-Zeros Substitution (T1 lines).
HDB3
High-Density Bipolar with 3-zeros (E1 lines).
AMI
Alternate Mark Inversion (E1 and T1 lines).
Make sure that the transmission encoding matches with the remote system. For further information,
see ITU-T Recommendation G.703.
7.
Select the frame format in the Line Framing drop-down menu.
Line Framing is used to synchronize the channels on the frame relay circuit (when a frame starts
and finishes). Without it, the sending and receiving equipment would not be able to synchronize
their frames.
Table 194: Line Framing
Format
Description
SF
Super frame. Sometimes known as D4 (T1 lines).
ESF
Extended super frame (T1 lines).
CRC4
Cyclic redundancy check 4 (E1 lines).
NO-CRC4 No Cyclic redundancy check 4 (E1 lines).
For further information, see ITU-T Recommendation G.704.
8.
Dgw v2.0 Application
Define the range of active bearer channels in the Channel Range field.
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Chapter 25 - E&M CAS Configuration
9.
E&M Channel Associated Signaling
Select the strategy for selecting bearer channels in the Channel Allocation Strategy drop-down
menu.
Table 195: Channel Allocation Strategy
Allocation
Description
Ascending
Starting from the lowest-numbered non-busy bearer channel and
going toward the highest-numbered non-busy bearer channel, the
Mediatrix unit selects the first bearer channel available.
Descending
Starting from the highest-numbered non-busy bearer channel and
going toward the lowest-numbered non-busy bearer channel, the
Mediatrix unit selects the first bearer channel available.
RoundRobinAscending
The Mediatrix unit starts from the bearer channel that follows the
bearer channel used for the last call. For instance, if channel #1 was
used in the last call, the unit starts with channel #2. Going toward the
highest-numbered non-busy bearer channel, the unit selects the first
channel available. If the highest channel is unavailable, the search
continues from the lowest-numbered non-busy bearer channel.
RoundRobinDescending The Mediatrix unit starts from the bearer channel that precedes the
bearer channel used for the last call. For instance, if channel #3 was
used in the last call, the unit starts with channel #2. Going toward the
lowest-numbered non-busy bearer channel, the unit selects the first
channel available. If the lowest channel is unavailable, the search
continues from the highest-numbered non-busy bearer channel.
10.
Define the maximum number of active calls on the interface in the Maximum Active Calls field.
This limits the total number of concurrent calls on the interface. Entering 0 indicates no maximum
number of active calls.
11.
Set the Encoding Scheme drop-down menu with the voice encoding scheme in the bearer
capabilities.
This encoding scheme is used when initiating a call on the E&M side. The supported encoding
schemes are G.711 u-Law and G.711 a-Law.
12.
Select the E&M Signaling Type Selection drop-down menu with the proper E&M signalling type.
Table 196: Signalling Type Selection
Signalling Type
Wink start
Description
Sets configuration parameters for basic Wink Start signalling. After
setting the SignallingType, individual configuration parameters can be
overriden for detail adjustment.
Note that the outgoing and incoming register signaling uses DTMF
signaling type by default.
Immediate Start
Sets configuration parameters for basic Immediate Start signalling.
After setting the Signaling Type, individual configuration parameters
can be overriden for detail adjustment.
Note that the outgoing and incoming register signaling uses DTMF
signaling type by default.
Feature Group B
Sets configuration parameters for the Feature Group B signalling
defined by National Exchange Carrier Association. After setting the
Signaling Type, individual configuration parameters can be overriden
for detail adjustment.
Note that the outgoing and incoming register signaling uses MF R1
signaling type by default.
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Software Configuration Guide
Table 196: Signalling Type Selection (Continued)
Signalling Type
Description
Feature Group D
Sets configuration parameters for the Feature Group D signalling
defined by National Exchange Carrier Association. After setting the
Signaling Type, individual configuration parameters can be overriden
for detail adjustment.
Note that the outgoing and incoming register signaling uses MF R1
signaling type by default.
13.
Set the Digit Attenuation field with the additional attenuation, in dB, for MFR1/DTMF digits
generation.
By default, MFR1/DTMF digits generation power is determined by variant selection. This parameter
provides a mean to reduce this power.
14.
Click Submit if you do not need to set other parameters.
E&M Signalling Variants
This section allows you to decide whether or not you want to override the default E&M signalling parameters.
The Mediatrix unit uses the following default values:
Table 197: E&M Signaling Parameters Default Values
Default Value (ms)
Parameter
Wink Start
FGBa
Immediate Start
FGDb
Bits BCD
8
8
8
8
ANI Length
0
0
0
0
DNIS Length
0
0
0
0
Incoming Register Signaling DTMF
DTMF
MfR 1
MfR 1
Outgoing Register Signaling DTMF
DTMF
MfR 1
MfR 1
Incoming Dial Map
%dnis%t
%dnis%t
%kp%dnis%st
%kp%ani%st%k
p%dnis%st
Outgoing Dial Map
%dnis%t
%dnis%t
%kp%dnis%st
%kp%dnis%st
Wait Wink
Enable
Disable
Enable
Enable
Wait Wink Ack
Disable
Disable
Disable
Enable
Send Wink
Enable
Disable
Enable
Enable
Send Wink Ack
Disable
Disable
Disable
Enable
a. Feature Group B
b. Feature Group D
Override Default Signaling Settings
You can override the default E&M signaling parameters. In that case, you will have access to the E&M
Signaling Variants section to define the signaling you want.
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Chapter 25 - E&M CAS Configuration
E&M Signalling Variants
 To override the E&M signaling default settings:
1.
In the web interface, click the E&M link, then the Signaling sub-link.
Figure 92: E&M Signaling Variants Web Page
2
3
2.
Select to which interface you want to apply the changes in the Select Interface drop-down menu at
the top of the window.
The number of interfaces available vary depending on the Mediatrix unit model you have.
3.
Select whether or not you want to override the default setting of E&M signalling parameters in the
Override Default Signaling Settings drop-down menu.
Table 198: E&M Signaling Override
Allocation
Description
Disable
The interface uses the default configuration associated with the
selected Signaling Type. The configuration set in the current row has
no effect on the default configuration of the selected Signaling Type.
Enable
The interface uses the specific signalling configuration as defined in
the E&M Signaling Variants section. To retrieve the default
configuration associated with the current Signaling Type, click the
Reset to Default button. Proceed to “E&M Signalling Variants” on
page 230.
Overriding the default settings is considered an advanced configuration. Media5 recommends not
to modify the signalling variants unless you know exactly what you are doing.
E&M Signalling Variants
This section allows you to define E&M signalling parameters. You can click the Reset to Default button at any
time to revert back to the default E&M signalling.
 To set E&M signalling parameters:
1.
In the E&M Signaling Variants section, set the B, C and D bits when the device transmits line signals
in the Bits BCD field.
The device ignores the B, C and D bits of received line signals.
Figure 93: E&M Signaling Variants Section
1
3
5
7
9
11
230
2
4
6
8
10
Dgw v2.0 Application
E&M Signalling Variants
Software Configuration Guide
BCD bits definition
•
0..7 : BCD bits are set to the specified value.
•
8 : BCD bits follow the A bit.
Note: On E1-CAS, the ABCD bit value cannot be set to 0000. If the BCD variable is set to 000 or to follow
the A bit, then the bitmask BCD=001 will be used instead.
2.
Set the ANI Length field with the expected length of Automatic Number Identification (ANI) to be
requested or sent.
ANI is the number of the calling party.
3.
•
0: Variable ANI length.
•
1..20: Specific ANI length.
Set the DNIS Length field with the expected length of the Dialed Number Identification Service
(DNIS).
DNIS is the called party or the destination number.
4.
•
0: Variable DNIS length used.
•
1..20: Specific DNIS length expected.
Set the Incoming Register Signaling drop-down menu with the proper incoming register signaling
method.
Table 199: Incoming Register Signalling Parameters
Parameter
5.
Description
MfR1
Multi Frequency - R1.
DTMF
Dual Tone Multi Frequency.
Set the Outgoing Register Signaling drop-down menu with the proper outgoing register signaling
method.
Table 200: Outgoing Register Signalling Parameters
Parameter
6.
Description
MfR1
Multi Frequency - R1.
DTMF
Dual Tone Multi Frequency.
Set the Incoming Dial Map field with the dial map expression to match against on the incoming side.
The dial map expression uses a special format using macros that enables you to construct a custom
dial map using the ANI, DNIS and other special tones as separator. The dial map macros supported
are:
•
%dnis : DNIS (Dialed Number Identification Service).
•
%ani : ANI (Automatic Number Identification).
•
%kp : KP (start-of-pulsing) tone (MF R1 only).
•
%st : ST (end-of-pulsing) tone (MF R1 only).
•
%t : Interdigit timeout.
•
A,B,C,D,*,# : Control tones (DTMF only).
•
A,B,C,D,E : Control tones (MF R1 only).
Examples:
Dgw v2.0 Application
•
%dnis*%ani : (dnis)*(ani)
•
%kp%dnis%st%kp%ani%st : KP(dnis)STKP(ani)ST
•
A%dnisBA%aniB : A(dnis)BA(ani)B
231
Chapter 25 - E&M CAS Configuration
7.
E&M Timers Variants
Set the Outgoing Dial Map field with the dial map format for the outgoing dial string.
The dial string generation uses a special format using macros that enables you to construct a
custom dial string using the ANI, DNIS and other specials tones used as separator. The dial string
uses regular expression to replace the macros with the proper call parameter value. The dial map
macros supported are:
•
%dnis : DNIS (Dialed Number Identification Service).
•
%ani : ANI (Automatic Number Identification).
•
%kp : KP (start-of-pulsing) tone (MF R1 only).
•
%st : ST (end-of-pulsing) tone (MF R1 only).
•
%t : Interdigit timeout.
•
A,B,C,D,*,# : Control tones (DTMF only).
•
A,B,C,D,E : Control tones (MF R1 only).
Examples : ANI=1234 DNIS=6789 KP=A ST=D
•
%dnis*%ani : 6789*1234
•
%kp%dnis%st%kp%ani%st : A6789DA1234D
•
A%dnisBA%aniB : A6789BA1234B
8.
Set the Wait Wink drop-down menu with whether or not the outgoing register should wait for a wink
before proceeding with digit transmission.
9.
Set the Wait Wink Ack drop-down menu with whether or not the outgoing register should wait for a
wink acknowledge after all digit reception.
10.
Set the Send Wink drop-down menu with whether or not the incoming register should send a wink
to notify the remote side that digit information can be sent.
11.
Set the Send Wink Ack drop-down menu with whether or not the incoming register should send a
wink to acknowledge the receipt of all digits.
12.
Click Submit if you do not need to set other parameters.
E&M Timers Variants
This section allows you to decide whether or not you want to override the default E&M timers parameters. The
Mediatrix unit uses the following default values:
Table 201: E&M Timers Default Values
Default Value (ms)
Parameter
Wink Start
232
FGBa
Immediate Start
FGDb
Backward Wait Pre Wink Timeout
50
50
50
50
Backward Send Wink Timeout
200
200
200
200
Backward Wait First Digit Timeout
10000
10000
10000
10000
Backward Clear Backward Timeout
2000
2000
2000
2000
Backward Digit Complete Timeout
4000
4000
4000
4000
Forward Wait Wink Timeout
5000
5000
5000
5000
Forward Wait Max Wink on Timeout 5000
5000
5000
5000
Forward Wait Pre Dial Timeout
140
140
140
140
Forward Wait Answer Timeout
180000
180000
180000
180000
Forward Clear Forward Timeout
2000
2000
2000
2000
Dgw v2.0 Application
E&M Timers Variants
Software Configuration Guide
Table 201: E&M Timers Default Values (Continued)
Default Value (ms)
Parameter
Wink Start
Release Guard Timeout
200
FGBa
Immediate Start
200
200
FGDb
200
a. Feature Group B
b. Feature Group D
Override Default Signaling Settings
You can override the default E&M timers. In that case, you will have access to the E&M Timers Variants
section to define the timers you want.
 To override the E&M timers default settings:
1.
In the web interface, click the E&M link, then the Timers sub-link.
Figure 94: E&M Timers Variants Web Page
2
3
2.
Select to which interface you want to apply the changes in the Select Interface drop-down menu at
the top of the window.
The number of interfaces available vary depending on the Mediatrix unit model you have.
3.
Select whether or not you want to override the default setting of E&M timers in the Override Default
Signaling Settings drop-down menu.
Table 202: E&M Timers Override
Allocation
Description
Disable
The interface uses the default configuration associated with the
selected Signaling Type. The configuration set in the current row has
no effect on the default configuration of the selected Signaling Type.
Enable
The interface uses the specific signaling configuration as defined in
the E&M Timers Variants section. To retrieve the default configuration
associated with the current signalling type, click the Reset to Default
button. Proceed to “E&M Timers Variants” on page 233.
Overriding the default settings is considered as advanced configuration. Media5 recommends not
to modify the signalling variants unless you know exactly what you are doing.
E&M Timers Variants
This section allows you to define E&M timers. You can click the Reset to Default button at any time to revert
back to the default E&M timers.
Dgw v2.0 Application
233
Chapter 25 - E&M CAS Configuration
E&M Digit Timers Variants
 To set E&M timers:
1.
In the E&M Timers Variants section, set the Backward Wait Pre Wink Timeout field with the amount
of time, in milliseconds (ms), an incoming register waits before sending the wink that acknowledges
the seizure.
Figure 95: E&M Timers Variants Section
1
3
5
2
4
6
7
9
11
8
10
2.
Set the Backward Send Wink Timeout field with the duration, in milliseconds (ms), of the wink signal
is applied by the incoming register to signal seizure acknowledgment.
3.
Set the Backward Wait First Digit Timeout field with the maximum time, in milliseconds (ms), an
incoming register waits for the first incoming digit after the line seizure.
4.
Set the Backward Clear Backward Timeout field with the maximum time, in milliseconds (ms), an
incoming register waits after sending a clear backward line signal before transiting to the idle state.
5.
Set the Backward Digit Complete Timeout field with the maximum time, in milliseconds (ms), the
incoming register waits for the next digit before considering the digit sequence as completed.
6.
Set the Forward Wait Wink Timeout field with the maximum time, in milliseconds (ms), an outgoing
register waits for seizure acknowledgement after seizing the line.
7.
Set the Forward Wait Max Wink On Timeout field with the maximum time, in milliseconds (ms), an
outgoing register waits for the seizure acknowledgment wink to complete.
8.
Set the Forward Wait Pre Dial Timeout field with the amount of time, in milliseconds (ms), an
outgoing register waits after the wink to start dialing.
9.
Set the Forward Wait Answer Timeout field with the maximum time, in milliseconds (ms), an
outgoing register waits for the call to be answered.
10.
Set the Forward Clear Forward Timeout field with the maximum time, in milliseconds (ms), an
outgoing register waits after sending a clear forward line signal before transiting to the idle state.
11.
Set the Release Guard Timeout field with the maximum time, in milliseconds (ms), a register waits
after sending an idle line signal to prevent a new seizure of the line.
12.
Click Submit if you do not need to set other parameters.
E&M Digit Timers Variants
This section allows you to decide whether or not you want to override the default E&M digit timers. The
Mediatrix unit uses the following default values:
Table 203: E&M Digit Timers Default Values
Default Value (ms)
Parameter
Wink Start
234
FGBa
Immediate Start
FGDb
KP On Timeout
100
100
100
100
KP Off Timeout
68
68
68
68
Dgw v2.0 Application
E&M Digit Timers Variants
Software Configuration Guide
a. Feature Group B
b. Feature Group D
Override Default Signaling Settings
You can override the default E&M digit timers. In that case, you will have access to the E&M Digit Timers
Variants section to define the timers you want.
 To override the E&M digit timers default settings:
1.
In the web interface, click the E&M link, then the Digit Timers sub-link.
Figure 96: E&M Digit Timers Variants Web Page
2
3
2.
Select to which interface you want to apply the changes in the Select Interface drop-down menu at
the top of the window.
The number of interfaces available vary depending on the Mediatrix unit model you have.
3.
Select whether or not you want to override the default setting of E&M digit timers in the Override
Default Signaling Settings drop-down menu.
Table 204: E&M Digit Timers Override
Allocation
Description
Disable
The interface uses the default configuration associated with the
selected Signaling Type. The configuration set in the current row has
no effect on the default configuration of the selected Signaling Type.
Enable
The interface uses the specific signaling configuration as defined in
the E&M Digit Timers Variants section. To retrieve the default
configuration associated with the current signalling, click the Reset to
Default button. Proceed to “E&M Digit Timers Variants” on page 235.
Overriding the default settings is considered as advanced configuration. Media5 recommends not
to modify the signalling variants unless you know exactly what you are doing.
E&M Digit Timers Variants
This section allows you to define E&M digit timers. You can click the Reset to Default button at any time to
revert back to the default E&M digit timers.
 To set E&M digit timers:
1.
In the E&M Digit Timers Variants section, set the KP On Timeout field with the time, in milliseconds
(ms), during which the MF R1 KP tone is on.
Figure 97: E&M Digit Timers Variants Section
1
2
Dgw v2.0 Application
235
Chapter 25 - E&M CAS Configuration
E&M Link Timers Variants
2.
Set the KP Off Timeout field with the duration, in milliseconds (ms), of the pause after the MF R1
KP tone.
3.
Click Submit if you do not need to set other parameters.
E&M Link Timers Variants
This section allows you to decide whether or not you want to override the default E&M link timers. The
Mediatrix unit uses the following default values:
Table 205: E&M Link Timers Default Values
Default Value (ms)
Parameter
Wink Start
FGBa
Immediate Start
FGDb
Link Activation Timeout
1000
1000
1000
1000
Link Activation Retry
Timeout
3000
3000
3000
3000
a. Feature Group B
b. Feature Group D
Override Default Signalling Settings
You can override the default E&M link timers. In that case, you will have access to the E&M Link Timers
Variants section to define the timers you want.
 To override the E&M link timers default settings:
1.
In the web interface, click the E&M link, then the Link Timers sub-link.
Figure 98: E&M Link Timers Variants Web Page
2
3
2.
Select to which interface you want to apply the changes in the Select Interface drop-down menu at
the top of the window.
The number of interfaces available vary depending on the Mediatrix unit model you have.
3.
Select whether or not you want to override the default setting of E&M link timers parameters in the
Override Default Signaling Settings drop-down menu.
Table 206: E&M Link Timers Override
Allocation
Disable
236
Description
The interface uses the default configuration associated with the
selected Signaling Type. The configuration set in the current row has
no effect on the default configuration of the selected Signaling Type.
Dgw v2.0 Application
E&M Tones Variants
Software Configuration Guide
Table 206: E&M Link Timers Override (Continued)
Allocation
Description
Enable
The interface uses the specific signaling configuration as defined in
the E&M Link Timers Variants section. To retrieve the default
configuration associated with the current signalling, click the Reset to
Default button. Proceed to “E&M Link Timers Variants” on page 237.
Overriding the default settings is considered as advanced configuration. Media5 recommends not
to modify the signalling variants unless you know exactly what you are doing.
E&M Link Timers Variants
This section allows you to define E&M link timers. You can click the Reset to Default button at any time to
revert back to the default E&M link timers.
 To set E&M link timers:
1.
In the E&M Link Timers Variants section, set the Link Activation Timeout field with the maximum
time, in milliseconds (ms), the unit waits for an activation indication coming from the physical link.
The activation indication is used to indicate that the physical layer connection has been activated.
Figure 99: R2 Link Timers Variants Section
1
2
2.
Set the Link Activation Retry Timeout field with the maximum time, in milliseconds (ms), the unit
waits before attempting to re-establish the physical link.
The attempt is made when the physical layer connection has been deactivated.
3.
Click Submit if you do not need to set other parameters.
E&M Tones Variants
This section allows you to decide whether or not you want to override the default E&M tones parameters. The
Mediatrix unit uses the following default values:
Table 207: E&M Tones Default Values
Default Value (ms)
Parameter
Wink Start
Immediate Start
FGBa
FGDb
KP Tone
MFC 10
MFC 10
MFC 10
MFC 10
ST Tone
MFC 13
MFC 13
MFC 13
MFC 13
a. Feature Group B
b. Feature Group D
Override Default Signalling Settings
You can override the default E&M tones. In that case, you will have access to the E&M Tones Variants section
to define the tone you want.
Dgw v2.0 Application
237
Chapter 25 - E&M CAS Configuration
E&M Tones Variants
 To override the E&M tones default settings:
1.
In the web interface, click the E&M link, then the Tones sub-link.
Figure 100: E&M Tones Variants Web Page
2
3
2.
Select to which interface you want to apply the changes in the Select Interface drop-down menu at
the top of the window.
The number of interfaces available vary depending on the Mediatrix unit model you have.
3.
Select whether or not you want to override the default setting of E&M tones parameters in the
Override Default Signaling Settings drop-down menu.
Table 208: E&M Tones Override
Allocation
Description
Disable
The interface uses the default configuration associated with the
selected Signaling Type. The configuration set in the current row has
no effect on the default configuration of the selected Signaling Type.
Enable
The interface uses the specific signalling configuration as defined in
the E&M Tones Variants section. To retrieve the default configuration
associated with the current signalling, click the Reset to Default
button. Proceed to “E&M Tones Variants” on page 238.
Overriding the default settings is considered as advanced configuration. Media5 recommends not
to modify the signalling variants unless you know exactly what you are doing.
E&M Tones Variants
This section allows you to define E&M tones. You can click the Reset to Default button at any time to revert
back to the default E&M tones.
 To set the E&M tones:
1.
In the E&M Tones Variants section, set the KP Tone drop-down menu with the proper KP (start-ofpulsing) signal.
You have the choice between none and MFR0 to MFR14 Tones.
Figure 101: E&M Tones Variants Section
1
2
2.
Set the ST Tone drop-down menu with the proper ST (end-of-pulsing) signal.
You have the choice between none and MFR0 to MFR14 Tones.
3.
238
Click Submit if you do not need to set other parameters.
Dgw v2.0 Application
PRI E&M Statistics
Software Configuration Guide
PRI E&M Statistics
This section describes configuration that is available only in the MIB parameters of the Mediatrix unit. You can
configure these parameters as follows:



by using a MIB browser
by using the CLI
by creating a configuration script containing the configuration variables
The Mediatrix unit collects meaningful statistics for each PRI digital card that can be read via SNMP and CLI.
The Mediatrix unit collects statistics for each of its two cards, if available. Slot 2 and Slot 3 indicate the physical
location of the cards in the unit, Slot 2 being on the left when looking at the rear of the unit.
 To view the PRI statistics:
1.
In the ex1Pri_1MIB, locate the statisticsGroup and expand it.
The following table describes the statistics available.
Table 209: E&M Statistics Displayed
Statistic
statsInfoFramesTra
nsmitted
Description
Number of HDLC frames transmitted.
Note: The term frames does not refer to the structure defined in I.431.
statsInfoFramesRec Number of HDLC frames received.
eived
Note: The term frames does not refer to the structure defined in I.431.
statsInfoOctetsTran
smitted
Number of octets transmitted. This value is obtained by cumulating the
octets transmitted in the HDLC frames.
Note: The term frames does not refer to the structure defined in I.431.
statsInfoOctetsRece Number of octets received. This value is obtained by cumulating the
ived
octets received in the HDLC frames.
Note: The term frames does not refer to the structure defined in I.431.
statsInfoFCSErrors
Frame check sequence (FCS) errors indicate that frames of data are
being corrupted during transmission. FCS error count is the number of
frames that were received with a bad checksum (CRC value) in the HDLC
frame. These frames are dropped and not propagated in the upper layers.
This value is available on E1 and T1.
statsInfoFramesDro
pped
Number of frames dropped. This value is obtained by cumulating the
number of frames dropped when transferring the data from the framer
chip to the device internal buffer.
This value is available on E1 and T1.
statsInfoOctetsDrop
ped
Number of octets dropped. This value is obtained by cumulating the
number of octets dropped when transferring the data from the framer chip
to the device internal buffer.
This value is available on E1 and T1.
statsInfoNegativeFr
ameSlipsTransmitte
d
A frame is skipped when the frequency of the transmit clock is greater
than the frequency of the transmit system interface working clock based
on 2.048 MHz (on E1) or 1.544 MHz (on T1).
This value is available on E1 and T1.
statsInfoNegativeFr
ameSlipsReceived
A frame is skipped when the frequency of the received route clock is
greater than the frequency of the receive system interface working clock
based on 2.048 MHz (on E1) or 1.544 MHz (on T1).
This value is available on E1 and T1.
Dgw v2.0 Application
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Chapter 25 - E&M CAS Configuration
PRI E&M Statistics
Table 209: E&M Statistics Displayed (Continued)
Statistic
Description
statsInfoPositiveFra
meSlipsTransmitted
A frame is repeated when the frequency of the transmit clock is less than
the frequency of the transmit system interface working clock based on
2.048 MHz (on E1) or 1.544 MHz (on T1).
This value is available on E1 and T1.
statsInfoPositiveFra
meSlipsReceived
A frame is repeated when the frequency of the receive route clock is less
than the frequency of the receive system interface working clock based
on 2.048 MHz (on E1) or 1.544 MHz (on T1).
This value is available on E1 and T1.
statsInfoFramingErr
or
The framing error count indicates that a FAS (Frame Alignment Signal)
word has been received with an error.
The FAS-bits are present in every even frame of timeslot 0 on E1.
The FAS-bits are present in ESF format on T1.
This value is available on E1 and T1.
statsInfoCodeViolati
ons
The code violations count indicates that an encoding error on the PCM
line has been detected.
This value is available on E1 and T1.
statsInfoCRCErrors
The CRC errors count is incremented when a multiframe has been
received with a CRC error.
The CRC error count is available in CRC multiframe mode only on E1.
The CRC error count is in ESF format on T1.
statsInfoE-BitError
The E-Bit error count gives information about the outgoing transmit PCM
line if the E-bits are used by the remote end for submultiframe error
indication. Incrementing is only possible in the multiframe synchronous
state.
Due to signaling requirements, the E-bits of frame 13 and frame 15 of the
CRC multiframe can be used to indicate an error in a received
submultiframes:
Submultiframe I status E-bit located in frame 13
Submultiframe II status E-bit located in frame 15
no CRC error : E = 1
CRC error : E = 0
This value is only available in E1.
statsInfoBlockError
The Block Error count is incremented once per multiframe if a multirame
has been received with a CRC error or a bad frame alignment has been
detected.
This value is only available for ESF format on T1 only.
 To reset the statistics:
1.
In the ex1Pri_1MIB, set the statsInfoResetStats variable to 10: resetStats.
You can also use the following line in the CLI or a configuration script:
ex1Pri_1.statsInfoResetStats=10
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Dgw v2.0 Application
SIP Parameters
Page Left Intentionally Blank
SIP Gateways Configuration
C
Software Configuration Guide
H A P T E R
26
SIP Gateways
This chapter describes how to add and remove SIP gateways in the Mediatrix unit.
SIP Gateways Configuration
Multiple SIP gateways may be used for a number of reasons, such as:


Redirecting ISDN calls to different SIP servers depending on the call.
Hunt calls across several gateways.
Adding a SIP gateway triggers a warning message if the total number of registrations configured reached the
defined limit. See “Number of Registrations” on page 259 for more details.
There are two types of SIP Gateways:
Trunk Gateway
A trunk gateway is generally used when the device is connected to a PBX or phone network; it can also be
used when connected to terminal equipment while using a SIP trunk to a SIP server.
Figure 102: Trunk Gateway
The characteristics of a trunk gateway are as follows:


It works with endpoint and user (unit) registrations.


A listening port allows for dialogs to be established by any peer.

Connections can be persistent or not, depending on the type of transport: UDP and TCP
transports are limited to non-persistent connections; TLS establishes persistent connections to
the outbound proxy, home domain proxy and registrar and non-persistent connections to other
targets.

The call router shows a single SIP source/destination for the gateway.
SIP dialogs are established independently of each other, depending on the SIP keep alive
mechanism defined. See “Keep Alive” on page 251 for more details on the SIP keep alive
mechanism.
When the destination is a FQDN, each SIP transaction is possibly sent to a different IP address,
depending on the DNS query result. A trunk gateway assumes all SIP servers identified by a
single FQDN have a synchronized state.
Endpoint Gateway
An endpoint gateway is generally used when the device is connected to terminal equipment where each
endpoint has a separate SIP connection to the SIP server.
Dgw v2.0 Application
243
Chapter 26 - SIP Gateways
SIP Gateways Configuration
Figure 103: Endpoint Gateway
An endpoint gateway is a type of gateway introduced to satisfy use cases with failover/failback based on
registrations for a single user.
The characteristics of an endpoint gateway are as follows:

It works with endpoint registrations only (no unit registrations can be associated to an endpoint
gateway).

SIP dialogs for a given SIP user can only be established once the user is registered to the
server.

It creates a persistent connection for each SIP user. This connection allows for dialogs to be
established only by the server to which the user is registered. “No listening port” allows a
connection to be established by a peer.

Failover/failback to another server requires the SIP user to register on that server prior to
establishing a dialog.

The call router and gateway status tables show an instance of the gateway for each user of the
gateway.
 To configure multiple SIP gateways:
1.
In the web interface, click the SIP link, then the Gateways sub-link.
Figure 104: SIP – Gateways Web Page
2
3
4
5
You can add a new gateway by clicking the
5 gateways.
6
button. The Mediatrix unit supports a maximum of
You can delete an existing gateway by clicking the
2.
7
button.
If you are adding a new gateway, enter its name in the Name field.
The Dgw v2.0 Application supports only alphanumeric characters, “-”, and “_”.
3.
Select the type of SIP gateway to be configured in the Type drop-down menu.
The default value is “Trunk”; select “Endpoint” for an endpoint gateway.
4.
Select the network interface on which the gateway listens for incoming SIP traffic in the Signaling
Network drop-down menu.
This value applies to all transports (e.g., UDP, TCP, etc.).
The LAN interface may be used as a SIP gateway to be bound on the LAN. However, there is no
routing between the LAN and the uplink interface.
244
Dgw v2.0 Application
SIP Gateways Configuration
Software Configuration Guide
5.
Define the list of networks (separated by ",") to use for the media (voice, fax, etc.) stream in the
Media Networks field.
You can use the Media Networks Suggestion column’s drop-down menu to select between
suggested values, if any.
The value must match one of the "InterarfaceName" values in the "NetworkInterfacesStatus" table
of the BNI service. The order in the list defines the priority.
When the media stream is negotiated, the following rules apply:
•
If the list of media networks is empty, the Mediatrix unit uses the IP address of the
network defined in the Signaling Network drop-down menu.
•
Only active networks are used.
•
Only the first active network of an IP address family (IPv4, IPv6) is used. All
subsequent networks of the same IP family are ignored.
Note: When generating an offer and multiple networks are available for the media, ANAT grouping (RFC
4091) is automatically enabled. When generating an answer, the ANAT grouping state is detected form the
offer.
6.
If the gateway type is set to "Trunk", set the SIP port on which the gateway listens for incoming
unsecure SIP traffic in the Port field.
This is used only when the UDP and/or TCP transports are enabled.
If two or more SIP gateways use the same port, only the first SIP gateway starts correctly. The
others are in error and not started. The SIP gateway is also in error and not started if the port is
already used.
The default value is 0. If you set the port to 0, the default SIP port 5060 is used.
Note: The port “0” is the equivalent to the “well known port”, which is 5060 in SIP. Using 0 and 5060 is not
the same. At the SIP packets level, if you set the port to 0, it will not be present in the SIP packet. If you set
the port to 5060, it will be present in the SIP packet. For example: “23@test.com” if the port is 0 and
“23@test.com:5060” if the port is 5060.
Note: When the gateway type is set to "Endpoint" the SIP Port and Secure Port have no effect and must
be set to 0 since each user has a unique UDP/TCP port.
For "Endpoint" gateways, only the base port can be set with the variable Persistent Base Port in the SIP
Transport configuration web interface.
7.
If the gateway type is set to "Trunk", set the SIP port on which the gateway listens for incoming
secure SIP traffic in the Secure Port field.
This is used only when the TLS transport is enabled.
The default value is 0. If you set the port to 0, the default secure SIP port 5061 is used.
Note: The port “0” is the equivalent to the “well known port”, which is 5061 in SIP for TLS. Using 0 and 5061
is not the same. At the SIP packets level, if you set the port to 0, it will not be present in the SIP packet. If
you set the port to 5061, it will be present in the SIP packet. For example: “23@test.com” if the port is 0 and
“23@test.com:5061” if the port is 5061.
8.
Click Submit if you do not need to set other parameters.
The state of the SIP gateways is displayed in the SIP Gateway Status section.
Table 210: SIP Gateway States
State
Ready
Dgw v2.0 Application
Description
The gateway is ready to make and receive calls.
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Table 210: SIP Gateway States
State
246
Description
Cannot start, port already in
use
The gateway cannot open its IP port because the port is
already used by another service. This generally occurs when
the administrator adds a new gateway but forgets to configure
a different IP port.
Network down
The SIP gateway is not started or the network interface on
which the SIP gateway is associated does not have an IP
address.
Restarting
The SIP gateway cannot make or receive calls while it is
restarting.
Waiting for time
synchronization
The gateway is started but it cannot open its SIP TLS port
because the real-time clock is not synchronized. This generally
occurs when the SNTP server is not set or is unreachable.
Server unreachable
The gateway is started but it cannot make and receive calls
because the SIP server is unreachable. This state is only
reported when a KeepAlive mechanism is used.
Unregistered
Indicates some registrations that are mandatory for this
gateway failed. See “Unregistered Unit Behaviour” on page 263
for more details.
Invalid Config
The gateway cannot start due to an inconsistent configuration.
Dgw v2.0 Application
Introduction
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Software Configuration Guide
H A P T E R
27
SIP Servers
This chapter describes how to configure the SIP server parameters of the Mediatrix unit.
It describes the following:


How to define the SIP servers IP information.
How to define the SIP gateways IP information.
Introduction
The Mediatrix unit uses the following types of servers:
Table 211: SIP Servers
Server
Description
Registrar Server
Accepts REGISTER requests and places the information it receives in those
requests into the location service for the domain it handles.
Proxy Server
An intermediary program that acts as both a server and a client for the purpose of
making requests on behalf of other clients. A proxy server primarily plays the role
of routing, which means its job is to ensure that a request is passed on to another
entity that can further process the request. Proxies are also useful for enforcing
policy and for firewall traversal. A proxy interprets, and, if necessary, rewrites
parts of a request message before forwarding it.
Outbound Proxy Server An intermediary entity that acts as both a server and a client for the purpose of
making requests on behalf of other clients. The outbound proxy receives all
outbound traffic and forwards it. Incoming traffic may or may not go through the
outbound proxy. The outbound proxy’s address is never used in the SIP packets,
it is only used as a physical network destination for the packets.
When the outbound proxy is enabled, the proxy is still used to create the To and
From headers, but the packets are physically sent to the outbound proxy.
Messaging Server Host A Messaging system host is a server that accepts MWI SUBSCRIBE requests
and places the information it receives in those requests into the location service
for the domain it handles.
SIP Outbound Proxy (From RFC 3261)
A proxy that receives requests from a client, even though it may not be the server resolved by the
Request-URI. Typically, a user agent is manually configured with an outbound proxy.
When enabled, the initial route for all SIP requests contains the outbound proxy address, suffixed with
the loose routing parameter “lr”. The Request-URI still contains the home domain proxy address.
Requests are directed to the first route (the outbound proxy).
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TLS Persistent Connections Status
TLS Persistent Connections Status
The TLS Persistent Connections Status table allows you to browse the status of the TLS persistent
connections of the Mediatrix unit. These connections are associated with the SIP servers (outbound proxy,
registrar and home domain proxy). Note that this section is not displayed if there is no information to show.
TLS connection is currently only supported with Trunk gateway.
Figure 105: SIP – TLS Persistent Connections Status Section
The following information is available:
Table 212: TLS Persistent Connection Parameters
Parameter
Description
Gateway
The SIP gateway used to register.
Local Port
Local port used by the TLS persistent connection.
Remote Host
The remote host used to establish the TLS persistent connection. The remote host
can be a host name or an IP address of the proxy, outbound proxy or registrar.
Remote IP Address The resolved IP address of the remote host used to establish the TLS persistent
connection.
Status
The current state of the TLS persistent connection.
•
Up: The TLS connection is established.
•
Down: The TLS connection is not established.
SIP Servers Configuration
This section describes how to configure the IP address and port number of the SIP servers.
If any of the SIP servers parameters corresponds to a domain name that is bound to a SRV record, the
corresponding port must be set to 0 for the unit to perform DNS requests of type SRV (as per RFC 3263).
Otherwise, the unit will not use DNS SRV requests, but will rather use only requests of type A because it does
not need to be specified which port to use.
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 To set the SIP servers configuration:
1.
In the web interface, click the SIP link, then the Configuration sub-link.
Figure 106: SIP – Servers Web Page
2
4
3
5
2.
Enter the SIP registrar server static IP address or domain name and port number in the Registrar
Host field.
You must enter the information as IP address:Port number. For instance:
192.168.0.5:5060
3.
Enter the SIP Proxy server static IP address or domain name and port number in the Proxy Host
field.
You must enter the information as IP address:Port number. For instance:
192.168.0.5:5060
4.
Enter the Messaging system host static IP address or domain name and port number in the
Messaging Server Host field.
If the host corresponds to a domain name that is bound to a SRV record, the port must be set to 0
for the unit to perform DNS SRV queries; otherwise only type A record lookups will be used.
You can define whether or not an endpoint needs to subscribe to a messaging system in “Endpoints
Registration” on page 255.
5.
Enter the SIP outbound proxy server static IP address or domain name and port number in the
Outbound Proxy Host field.
The outbound proxy is enabled if the IP address is valid (i.e., not 0.0.0.0:0). Setting the address to
0.0.0.0:0 or leaving the field empty disables the outbound proxy.
Note: The Endpoint gateway can only have a single next hop. If no outbound proxy is set then, if used, the
proxy host, the registrar host and the messaging server host must be set to the same FQDN or IP Address.
If the hosts are not set to the same URL or IP address, the SIP Gateway State will be set to Invalid Config.
See Table 210 on page 245 for the list of SIP Gateway states.
6.
Click Submit if you do not need to set other parameters.
Multiple SIP Gateways
The Mediatrix unit allows you to have multiple SIP gateways (interfaces). You can configure each SIP gateway
to register to a specific registrar. You can also configure each SIP gateway to send all requests to an outbound
proxy. See “Chapter 26 - SIP Gateways” on page 243 for more details.
SIP Gateway Specific Registrar Servers
This section allows you to define whether the available SIP gateways use the default registrar server or rather
use a specific registrar server.
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 To set specific registrars servers information:
1.
In the Registrar Servers section of the Servers page, select whether or not a SIP gateway uses a
specific registrar server in the Gateway Specific drop-down menu.
If you select No, the SIP gateway uses the server information as set in the SIP Default Servers
section.
Figure 107: SIP Servers – Specific Registrar Section
1
2.
2
Enter the IP address or domain name and port number of the registrar server currently used by the
registration in the Registrar Host field.
You must enter the information as IP address:Port number. For instance:
192.168.0.5:5060
3.
If you do not need to set other parameters, do one of the following:
•
To save your settings without refreshing the registration, click Submit.
•
To save your settings and refresh the registration now, click Submit & Refresh
Registration.
SIP Gateway Specific Messaging Servers
This section allows you to define whether the available SIP gateways use the default proxy and outbound
proxy server or rather use specific servers.
 To set specific proxy servers information:
1.
In the Messaging Servers section of the Servers page, select whether or not a SIP gateway uses a
specific proxy and outbound proxy server in the Gateway Specific drop-down menu.
If you select No, the SIP gateway uses the server information as set in the SIP Default Servers and
Messaging Subscription (“Messaging Subscription” on page 317) sections.
Figure 108: SIP Servers – Messaging Section
1
2.
2
Enter the IP address or domain name and port number of the messaging server currently used by
the registration in the Proxy Host field.
You must enter the information as IP address:Port number. For instance:
192.168.0.5:5060
3.
Enter the IP address or domain name and port number of the outbound proxy server currently used
by the registration in the Outbound Proxy Host field.
You must enter the information as IP address:Port number. For instance:
192.168.0.5:5060
The outbound proxy is enabled if the IP address is valid (i.e., not 0.0.0.0:0). Setting the address to
0.0.0.0:0 or leaving the field empty disables the outbound proxy.
4.
250
If you do not need to set other parameters, do one of the following:
•
To save your settings without refreshing the registration, click Submit.
•
To save your settings and refresh the registration now, click Submit & Refresh
Registration.
Dgw v2.0 Application
Multiple SIP Gateways
Software Configuration Guide
SIP Gateway Specific Proxy Servers
This section allows you to define whether the available SIP gateways use the default proxy and outbound
proxy server or rather use specific servers.
 To set specific proxy servers information:
1.
In the Proxy Servers section of the Servers page, select whether or not a SIP gateway uses a
specific proxy and outbound proxy server in the Gateway Specific drop-down menu.
If you select No, the SIP gateway uses the server information as set in the SIP Default Servers
section.
Figure 109: SIP Servers – Specific Proxy Section
1
2.
2
3
Enter the IP address or domain name and port number of the proxy server currently used by the
registration in the Proxy Host field.
You must enter the information as IP address:Port number. For instance:
192.168.0.5:5060
Note: The Endpoint gateway can only have a single next hop. If no outbound proxy is set then, if used, the
proxy host, the registrar host and the messaging server host must be set to the same FQDN or IP Address.
If the hosts are not set to the same URL or IP address, the SIP Gateway State will be set to Invalid Config.
See Table 210 on page 245 for the list of SIP Gateway states.
3.
Enter the IP address or domain name and port number of the outbound proxy server currently used
by the registration in the Outbound Proxy Host field.
You must enter the information as IP address:Port number. For instance:
192.168.0.5:5060
The outbound proxy is enabled if the IP address is valid (i.e., not 0.0.0.0:0). Setting the address to
0.0.0.0:0 or leaving the field empty disables the outbound proxy.
4.
If you do not need to set other parameters, do one of the following:
•
To save your settings without refreshing the registration, click Submit.
•
To save your settings and refresh the registration now, click Submit & Refresh
Registration.
Keep Alive
You can select the method used to perform the SIP keep alive mechanism. With this mechanism, the Mediatrix
unit sends messages periodically to the server to ensure that it can still be reached. The SIP keep alive
mechanism is only supported with Trunk gateways.
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 To use the SIP keep alive mechanism:
1.
In the Keep Alive section of the Servers page, select the keep alive method to use in the Keep Alive
Method drop-down menu.
Figure 110: Keep Alive Section
1
2
3
Table 213: Keep Alive Parameters
Parameter
Description
None
No keep alive is performed.
SipOptions
SIP OPTIONS are sent periodically for each gateway to the
corresponding server. Any response received from the
server means that it can be reached. No additional
processing is performed on the response. If no response is
received after the retransmission timer expires
(configurable via the Transmission Timeout field in “SIP
Interop” on page 279), the gateway considers the server as
unreachable. In this case, any call attempt through the
gateway is refused. SIP OPTIONS are still sent when the
server cannot be reached and as soon as it can be reached
again, new calls are allowed.
Ping
A Ping is sent periodically for each gateway to the
corresponding server. The response received from the
server means that it is reachable. If no response is received
after the retransmission timer expires (configurable via the
Transmission Timeout field in “SIP Interop” on page 279),
the gateway considers the server as unreachable. In this
case, any call attempt through the gateway is refused. The
Pings are still sent when the server is unreachable and as
soon as it becomes reachable again, new calls are allowed.
On Endpoint gateways, the keep alive mechanism is always considered to be “None”.
2.
Set the interval, in seconds, at which SIP Keep Alive requests using SIP OPTIONS or Ping are sent
to verify the server status in the Keep Alive Interval field.
3.
Select the behaviour of the device when performing the keep alive action in the Keep Alive
Destination drop-down menu.
Table 214: SIP Keep Alive Destination Parameters
Parameter
4.
252
Description
First SIP Destination
Performs the keep alive action through the first SIP
destination. This corresponds to the outbound proxy host
when specified, otherwise it is the proxy host.
Alternate Destination
Performs the keep alive action through the alternate
destination target (see “SIP Gateway Specific Keep Alive
Destinations” on page 253 for more details).
Click Submit if you do not need to set other parameters.
Dgw v2.0 Application
Outbound Proxy Loose Router Configuration
Software Configuration Guide
SIP Gateway Specific Keep Alive Destinations
This section allows you to override the default Keep Alive destination alternate target when the Keep Alive
Destination drop-down menu is set to Alternate Destination (see “Keep Alive” on page 251 for more details).
 To set specific keep alive destinations:
1.
In the Keep Alive Destinations section of the Servers page, set the Alternate destination target
server FQDN and port for a specific SIP gateway in the default field.
You must enter the information as IP address:Port number. For instance:
192.168.0.5:5060
Figure 111: SIP Servers – Specific Keep alive Targets
1
2.
Click Submit if you do not need to set other parameters.
Outbound Proxy Loose Router Configuration
This section describes configuration that is available only in the MIB parameters of the Mediatrix unit. You can
configure these parameters as follows:



by using a MIB browser
by using the CLI
by creating a configuration script containing the configuration variables
You can specify the type of routing of the outbound proxy configured in “SIP Servers Configuration” on
page 248.
You can use two types of configuration:


Default configurations that apply to all the endpoints of the Mediatrix unit.
Specific configurations that override the default configurations.
You can define specific configurations for each endpoint in your Mediatrix unit. For instance, you
could enable a codec for all the endpoints of the Mediatrix unit and use the specific configuration
parameters to disable this same codec on one specific endpoint.
Using one or more specific parameter usually requires that you enable an override variable and set the specific
configuration you want to apply.
The following types are available:
Table 215: Outbound Proxy Router Status
Type
Description
LooseRouter This is the most current method for SIP routing, as per RFC 3261, and will become the
standard behaviour once RFC 3261 compliance is achieved. See “Introduction” on
page 247 for details.
Loose Router
A proxy is said to be loose routing if it follows the procedures defined in the RFC 3261 specification
(section 6) for processing of the Route header field. These procedures separate the destination of the
request (present in the Request-URI) from the set of proxies that need to be visited along the way
(present in the Route header field). A proxy compliant to these mechanisms is also known as a loose
router.
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Table 215: Outbound Proxy Router Status (Continued)
Type
Description
StrictRouter
Pre-RFC 3261, RFC 2543 compatible SIP routing.
The initial route for all SIP requests contains the home domain proxy address (the RequestURI). Requests are directed to the outbound proxy.
In other words, the Request-URI is constructed as usual, using the home domain proxy and
the user name, but is used in the route set. The Request-URI is filled with the outbound
proxy address.
NoRouteHea Removes the route header from all SIP packets sent to an outbound proxy. This does not
der
modify persistent TLS connection headers.
Note: The Router header will not be removed from the SIP packets if the unit is configured
to use the TLS Fallback feature. This feature requires the information of the SIP Outbound
Proxy in the SIP packet to work correctly.
 To set the outbound proxy router status:
1.
In the sipEpMIB, set the defaultProxyOutboundType variable to the proper value.
You can also use the following line in the CLI or a configuration script:
sipEp.defaultProxyOutboundType="Value"
where Value may be one of the following:
Table 216: Outbound Proxy Router Values
Value
2.
Meaning
100
LooseRouter
200
StrictRouter
300
NoRouteHeader
If you want to set a different routing type for one or more SIP gateways, set the following variables:
•
gwSpecificproxyEnableConfig variable for the specific SIP gateway you want to
configure to enable.
•
gwSpecificProxyOutboundType variable for the specific SIP gateway you want to
configure to the proper value.
You can also use the following lines in the CLI or a configuration script:
sipEp.gwSpecificProxy.EnableConfig[GatewayName="default"]="1"
sipEp.gwSpecificProxy.OutboundType[GatewayName="Specific_Gateway"]="Value"
where:
254
•
Specific_Gateway is the name of the SIP gateway you want to configure.
•
Value is the refresh router status as defined in Step 1.
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Endpoints Registration
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Software Configuration Guide
H A P T E R
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SIP Registration
This chapter describes how to configure the registration parameters of the Mediatrix unit.
Endpoints Registration
Each endpoint of the Mediatrix unit has its own registration information. You can set information for each
endpoint such as its telephone number and friendly name.
Adding an endpoint registration triggers a warning message if the total number of registrations configured
reached the defined limit. See “Number of Registrations” on page 259 for more details.
 To set endpoints registration information:
1.
In the web interface, click the SIP link, then the Registrations sub-link.
Figure 112: SIP – Registrations Web Page
2
2.
3
4
5
6
In the Endpoints Registration and Subscription section of the Registrations page, enter a user name
for each endpoint in the User Name column.
The user name (such as a telephone number) uniquely identifies this endpoint in the domain. It is
used to create the Contact and From headers. The From header carries the permanent location (IP
address, home domain) where the endpoint is located. The Contact header carries the current
location (IP address) where the endpoint can be reached.
Contacts are registered to the registrar. This enables callers to be redirected to the endpoint’s
current location.
Note: If two or more endpoints have the same user name, a single registration request and/or subscription
request will be performed under that user name.
3.
Enter another name for each endpoint in the Friendly Name column.
This is a friendly name for the endpoint. It contains a descriptive version of the URI and is intended
to be displayed to a user interface.
4.
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Define whether or not the endpoint registration needs to register to the registrar in the Register
column.
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Endpoints Registration
An endpoint configured to register (set to Enable) will become unavailable for calls from or to SIP
when not registered.
You can define the behaviour of an endpoint when it becomes unavailable in the
defaultRegistrationUnregisteredBehavior MIB variable.
5.
Define whether or not the endpoint needs to subscribe to a messaging system in the Messaging
drop-down menu.
The current state of the subscription is displayed in the Endpoints Messaging Subscription Status
table.
Table 217: MWI Subscription State
State
Description
Unsubscribed
The unit/endpoint is not subscribed and never tries to subscribe. This case
occurs if the network interface used by the SIP gateway is not up or the unit/
endpoint is locked.
Subscribing
The subscription is currently trying to subscribe.
Subscribed
The subscription is successfully subscribed.
Refreshing
The subscription is trying to refresh.
Unreachable
The last subscription attempt failed because the messaging server is
unreachable.
AuthFailed
The last subscription attempt failed because authentication was not
successful.
Rejected
The last subscription attempt failed because the messaging server rejects
the subscription.
ConfigError
The last subscription attempt failed because it was badly configured. Check
if the username and the messaging host are not empty.
InvalidResponse The received 200 OK response contact does not match the contact of the
messaging server, or the 200 OK response for an unsubscribe contains a
contact.
You can enter the address of the Messaging server in “SIP Servers Configuration” on page 248.
6.
Select on which SIP gateway the user configuration is applied in the Gateway Name drop-down
menu.
You must have SIP gateways already defined. See “Chapter 26 - SIP Gateways” on page 243 for
more details. If you select all, the configuration applies to all gateways available.
7.
If you do not need to set other parameters, do one of the following:
•
To save your settings without refreshing the registration, click Submit.
•
To save your settings and refresh the registration now, click Submit & Refresh.
Contact Domain
This section describes configuration that is available only in the MIB parameters of the Mediatrix unit. You can
configure these parameters as follows:



by using a MIB browser
by using the CLI
by creating a configuration script containing the configuration variables
You can set the host part of the SIP contact field. If an empty string is specified, the listening IP address is
used.
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 To set the contact domain:
1.
In the sipEp MIB, set the UserAgentContactDomain variable in the UserAgent table or,
2.
In the CLI or a configuration script, use:
SipEp.UserAgent[EpId=index).ContactDomain=value
Accept Language
The AcceptLanguage parameter allows a user to indicate the preferred language that will be used for
displayed phrases, session descriptions, status responses carried as message bodies in the response. It is
used to fill the Accept-Language SIP header field.
You can configure the parameter by:



using a MIB browser
using the CLI
creating a configuration script containing the configuration variables
 To set AcceptLanguage:
1.
In the SipEp Mib. set the AcceptLanguage variable in UserAgent table or,
2.
In the CLI or a configuration script use
SipEp.UserAgent[index-value].AcceptLanguage=<value>
Index is the endpoint name
Unit Registration
Unit registration is used to register a contact not directly related to endpoints. This is generally used to indicate
to a registrar the IP location of the Mediatrix unit when it is used as a gateway.
Adding a unit registration triggers a warning message if the total number of registrations configured reached
the defined limit. See “Number of Registrations” on page 259 for more details.
 To set unit registration information:
1.
In the Unit Registration section of the Registrations page, enter a user name in the User Name
column.
Figure 113: SIP Registrations – Unit Registration Section
1
2
The user name (such as a telephone number) uniquely identifies this user in the domain.
You can add a new user by clicking the
button.
You can delete an existing user by clicking the
2.
button.
Select on which SIP gateway the user configuration is applied in the Gateway Name drop-down
menu.
You must have SIP gateways already defined. See “Chapter 26 - SIP Gateways” on page 243 for
more details. If you select all, the configuration applies to all gateways available.
3.
If you do not need to set other parameters, do one of the following:
•
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To save your settings without refreshing the registration, click Submit.
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Registration Configuration
•
To save your settings and refresh the registration now, click Submit & Refresh.
Registration Configuration
This section allows you to define registration refresh parameters.
See “Additional Registration Refresh Parameters” on page 260 for more registration parameters.
 To set the registration configuration:
1.
In the Registration Configuration section of the Registrations page, set the Default Registration
Refresh Time field with the time, in seconds, at which a registered unit begins updating its
registration before the registration expiration.
Figure 114: SIP Registrations – Registration Configuration Section
1
3
2
In SIP, a registration is valid for a period of time defined by the registrar. Once a unit is registered,
the SIP protocol requires the User Agent to refresh this registration before the registration expires.
Typically, this re-registration must be completed before the ongoing registration expires, so that the
User Agent's registration state does not change (i.e., remains 'registered').
For instance, if the parameter is set to 43 and the registration lasts one hour, the unit will send new
REGISTER requests 59 minutes and 17 seconds after receiving the registration acknowledgement
(43 seconds before the unit becomes unregistered).
Note: Normally, the Mediatrix unit cannot make or receive calls until the REGISTER has completed
successfully. Because the timeout for a SIP transaction in UDP is 32 seconds, it is possible to have an
ongoing re-REGISTER transaction at the same moment that the registration itself expires. This could
happen if the Default Registration Refresh Time field is set to a value lower than 32.
In that case, the user agent becomes unregistered, and will become registered again only when the reREGISTER request is answered with a positive response from the server. See “Gateway Specific
Registration Retry Time” on page 262 for a workaround if the unit cannot make calls during that period.
Setting this parameter to 0 means that the User Agent will fall into the 'unregistered' state BEFORE
sending the re-REGISTER requests.
This value MUST be lower than the value of the "expires" of the contact in the 200 OK response to
the REGISTER, otherwise the unit rapidly sends REGISTER requests continuously.
You can also set a different registration refresh time for one or more SIP gateways by using the MIB
parameters of the Mediatrix unit. See “Registration Refresh” on page 261 for more details.
2.
Set the Proposed Expiration Value In Registration field with the suggested expiration delay, in
seconds, of a contact in the REGISTER request.
The SIP protocol allows an entity to specify the “expires” parameter of a contact in a REGISTER
request. The server can return this “expires” parameter in the 200 OK response or select another
“expires”. In the REGISTER request, the “expires” is a suggestion the entity makes.
The “expires” parameter indicates how long, in seconds, the user agent would like the binding to be
valid.
Available values are from 1 s to 86,400 s (one day).
This value does not modify the delay before a re-REGISTER.
258
•
The delay is the “expires” of the contact in the 200 OK response to the REGISTER
request minus the value set in the Default Registration Refresh Time field.
•
If the “expires” of the contact in the 200 OK response to the REGISTER is not present
or not properly formatted, then the delay is the default registration proposed expiration
value minus the value set in the Default Registration Refresh Time field.
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Setting the parameter to 0 disables the expiration suggestion.
You can also set a different expiration delay for one or more SIP gateways by using the MIB
parameters of the Mediatrix unit. See “Registration Expiration” on page 261 for more details.
3.
Set the Default Expiration Value in Registration field with the default registration expiration, in
seconds.
This value is used when the contact in a registration response contains no “expires” or the “expires”
is badly formatted. In this case, the delay before a re-REGISTER is the value set in this field minus
the value set in the Default Registration Refresh Time field (Step 1).
You can also set a different expiration value in registration for one or more SIP gateways by using
the MIB parameters of the Mediatrix unit. See “Expiration Value in Registration” on page 261 for
more details.
4.
If you do not need to set other parameters, do one of the following:
•
To save your settings without refreshing the registration, click Submit.
•
To save your settings and refresh the registration now, click Submit & Refresh.
Number of Registrations
The Mediatrix unit limits the total number of registrations to 100. The total number of registrations is the sum
of all the endpoints and gateways (“SIP Gateways Configuration” on page 243) pairs. The Mediatrix unit
supports a maximum of 5 gateways. An endpoint configured with "All" gateways generates as many pairs as
the number of gateways. In a setup with 3 gateways, one endpoint configured with "All" as the gateway name
counts for 3 in the total number of registrations.
The registrations are enabled gateway by gateway until the limit is reached. Endpoints Registrations are used
first, then Unit Registrations. The remaining registrations are not registered and do not appear in the status
table. If you click the Submit And Refresh button and the configured number of registrations exceeds the
defined limit, a warning is displayed on the web interface (as well as in the CLI and SNMP interfaces) and a
syslog notify (Level Error) is sent.
Adding a gateway or an endpoint triggers a warning message if the total number of registrations configured
reached the defined limit.
Let’s suppose for instance that we have the current SIP Gateways configuration and the following SIP
Registration configuration:
Figure 115: Example, Gateway Configuration
Figure 116: Example, Registrations Configuration
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The following table describes how to compute the total number of registrations for this example:
Table 218: Number of Registrations Example
Parameter
Setting
Nb of Registrations
Endpoint Registration 1 in
Figure 116
Gateway Name set to alla
3
Endpoint Registration 2 in
Figure 116
Gateway Name set to gw2
1
Unit Registration 1 in Figure 116
Gateway Name set to all
3
Unit Registration 2 in Figure 116
Gateway Name set to all
3
Unit Registration 3 in Figure 116
Gateway Name set to gw1
1
Unit Registration 4 in Figure 116
Gateway Name set to default 1
Total Number of registrations
12
a. When the Gateway Name is set to all, this must be multiplied by the number of gateways set in Figure 115.
In this example, there are 3 gateways set.
Additional Registration Refresh Parameters
This section describes configuration that is available only in the MIB parameters of the Mediatrix unit. You can
configure these parameters as follows:



by using a MIB browser
by using the CLI
by creating a configuration script containing the configuration variables
Default Registration Retry Time
You can configure the interval in seconds (s) on which a failed registration is retried.
This variable defines the time, relative to the failure of the registration, at which the device retries the
registration.
 To specify the default registration retry time value:
1.
In the sipEpMIB, locate the registrationGroup folder.
2.
Set the DefaultRegistrationRetryTime variable with the desired interval value.
You can also use the following line in the CLI or a configuration script:
sipEp.DefaultRegistrationRetryTime="Value"
where Value may be between 1 and 86400 seconds.
Default vs. Specific Configurations
You can use two types of configuration:


Default configurations that apply to all the endpoints of the Mediatrix unit.
Specific configurations that override the default configurations.
You can define specific configurations for each endpoint in your Mediatrix unit. For instance, you
could enable a codec for all the endpoints of the Mediatrix unit and use the specific configuration
parameters to disable this same codec on one specific endpoint.
Using one or more specific parameter usually requires that you enable an override variable and set the specific
configuration you want to apply.
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Registration Refresh
You can set the default registration refresh time in the web page (“Registration Configuration” on page 258),
but you can also set a different registration refresh time for one or more SIP gateways.
 To set registration refresh parameters:
1.
In the sipEpMIB, locate the registrationGroup folder.
2.
If you want to set a different registration refresh time for one or more SIP gateways, set the following
variables:
•
gwSpecificRegistrationEnableConfig variable for the specific SIP gateway you
want to configure to enable.
•
gwSpecificRegistrationRefreshTime variable for the specific SIP gateway you want
to configure to the proper value.
You can also use the following lines in the CLI or a configuration script:
sipEp.gwSpecificRegistration.EnableConfig[GatewayName="Specific_Gateway"]="1"
sipEp.gwSpecificRegistration.RefreshTime[GatewayName="Specific_Gateway"]="Value"
where:
•
Specific_Gateway is the name of the SIP gateway you want to configure.
•
Value is the refresh time value.
Registration Expiration
You can set the default registration proposed expiration value in the web page (“Registration Configuration”
on page 258), but you can also set a different registration refresh time for one or more SIP gateways.
 To configure the registration expiration:
1.
In the sipEpMIB, locate the registrationGroup folder.
2.
If you want to set a different registration refresh time for one or more SIP gateways, set the following
variables:
•
gwSpecificRegistrationEnableConfig variable for the specific SIP gateway you
want to configure to enable.
•
gwSpecificRegistrationProposedExpirationValue variable for the specific SIP
gateway you want to configure to the proper value.
You can also use the following lines in the CLI or a configuration script:
sipEp.gwSpecificRegistration.EnableConfig[GatewayName="Specific_Gateway"]="1"
sipEp.gwSpecificRegistration.ProposedExpirationValue[GatewayName="Specific_Gatew
ay"]="Value"
where:
•
Specific_Gateway is the name of the SIP gateway you want to configure.
•
Value is the expiration delay value.
This value does not modify the time before a re-REGISTER.
•
The delay is the “expires” of the contact in the 200 OK response to the REGISTER
request minus the value set in the gwSpecificRegistrationRefreshTime parameter.
•
If the “expires” of the contact in the 200 OK response to the REGISTER is not present
or not properly formatted, then the delay is the default registration proposed expiration
value minus the value set in the gwSpecificRegistrationRefreshTime parameter.
Expiration Value in Registration
You can set the default expiration value in registration in the web page (“Registration Configuration” on
page 258), but you can also set a different expiration value in registration for one or more SIP gateways.
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This value is used when the contact in a registration response contains no “expires” or the “expires” is badly
formatted. In this case, the delay before a re-REGISTER is the value set in this field minus the value set in the
in the 'RefreshTime' variable (“Registration Refresh” on page 261).
 To configure the expiration value in registration for a specific gateway:
1.
In the sipEpMIB, locate the registrationGroup folder.
2.
To set a different expiration value in registration for one or more SIP gateways, set the following
variables:
•
gwSpecificRegistrationEnableConfig variable for the specific SIP gateway you
want to configure to enable.
•
gwSpecificRegistrationExpirationValue variable for the specific SIP gateway you
want to configure to the proper value.
You can also use the following lines in the CLI or a configuration script:
3.
To set a different expiration value in registration for one or more SIP gateways, put the following
lines in the configuration script:
sipEp.gwSpecificRegistration.EnableConfig[GatewayName="Specific_Gateway"]="1"
sipEp.gwSpecificRegistration.ExpirationValue[GatewayName="Specific_Gateway"]="Va
lue"
where:
•
Specific_Gateway is the name of the SIP gateway you want to configure.
•
Value is the expiration value in registration value.
Gateway Specific Registration Retry Time
You can set a different Registration Retry Time for one or more SIP gateways.
This variable defines the time, relative to the failure of the registration, at which the SIP gateway retries the
registration.
 To specify the registration retry time value for a specific gateway:
1.
In the sipEpMIB, locate the registrationGroup folder.
2.
To set a different registration retry time for one or more SIP gateways, set the following variables:
•
gwSpecificRegistrationEnableConfig variable for the specific SIP gateway you
want to configure to enable.
•
gwSpecificRegistrationRetryTime variable for the specific SIP gateway you want to
configure to the proper value.
You can also use the following line in the CLI or a configuration script:
3.
To set a different expiration value in registration for one or more SIP gateways, put the following
lines in the configuration script:
sipEp.gwSpecificRegistration.EnableConfig[GatewayName="Specific_Gateway"]="1"
sipEp.gwSpecificRegistrationRetryTime[GatewayName="Specific_Gateway"]="Value"
where:
•
Specific_Gateway is the name of the SIP gateway you want to configure.
•
Value is the expiration value in registration retry time.
Unregistered Endpoint Behaviour
You can specify whether an endpoint should remain enabled or not when not registered. This is useful if you
want your users to be able to make calls even if the endpoint is not registered with a SIP server.
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The following values are supported:
Table 219: Unregistered Endpoint Behaviour Parameters
Value
Description
disablePort
When the endpoint is not registered, it is disabled. The user cannot make or receive calls.
Picking up the handset yields a fast busy tone, and incoming INVITEs receive a “403
Forbidden” response.
enablePort
When the endpoint is not registered, it is still enabled. The user can receive and initiate
outgoing calls. Note that because the endpoint is not registered with a registrar, its public
address is not available to the outside world; the endpoint will most likely be unreachable
except through direct IP calling.
 To specify unregistered endpoint behaviour:
1.
In the sipEpMIB, locate the registrationGroup folder.
2.
Set the defaultRegistrationUnregisteredBehavior variable.
You can also use the following line in the CLI or a configuration script:
sipEp.defaultRegistrationUnregisteredBehavior="Value"
where Value may be as follows:.
Table 220: Unregistered Endpoint Behaviour Values
Value
3.
Meaning
0
disablePort
1
enablePort
If you want to set a different behaviour for one or more SIP gateways, set the following variables:
•
gwSpecificRegistrationEnableConfig variable for the specific SIP gateway you
want to configure to enable.
•
gwSpecificRegistrationUnregisteredBehavior variable for the specific SIP
gateway you want to configure to the proper value.
You can also use the following lines in the CLI or a configuration script:
sipEp.gwSpecificRegistration.EnableConfig[GatewayName="Specific_Gateway"]="1"
sipEp.gwSpecificRegistration.UnregisteredBehavior[GatewayName="Specific_Gateway"
]="Value"
where:
•
Specific_Gateway is the name of the SIP gateway you want to configure.
•
Value is one of the values described in Step 2.
Unregistered Unit Behaviour
You can specify whether the SIP gateway state should be affected or not by the unit registrations state.
The following values are supported:
Table 221: Unregistered Unit Behaviour Parameters
Value
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Description
NoEffect
The unit registrations state has no effect on the SIP gateway state.
DisableGate
way
The SIP gateway goes in the 'unregistered' state when all unit registrations are not in the
'registered' state. The 'unregistered' state indicates some registrations that are mandatory
for this gateway failed.
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 To specify unregistered unit behaviour:
1.
In the sipEpMIB, locate the registrationGroup folder.
2.
Set the defaultUnitRegistrationUnregisteredBehavior variable.
You can also use the following line in the CLI or a configuration script:
sipEp.defaultUnitRegistrationUnregisteredBehavior="Value"
where Value may be as follows:.
Table 222: Unregistered Unit Behaviour Values
Value
Meaning
100
NoEffect
200
DisableGateway
Behaviour on Initial-Registration Reception
You can configure the behaviour of the Mediatrix unit upon reception of a 380 or 504 carrying an XML body
with a specified 'initial-registration' action.
The following values are supported:
Table 223: Behaviour on Initial-Registration Reception Parameters
Value
Description
NoRegistration
No registration refresh are sent upon reception of the message.
EndpointRegistration
Registration refresh of the endpoint associated with the call is sent upon
reception of the message.
UnitRegistration
Registration refresh of all the usernames configured as 'unit registration'
are sent upon reception of the message. When there are duplicates, only
one REGISTER request is sent for all the duplicates.
UnitAndEndpointRegistration Registration refresh of the endpoint associated with the call and of all the
usernames configured as 'unit registration' are sent upon reception of the
message.
 To specify the behaviour on Initial-Registration reception:
1.
In the sipEpMIB, locate the registrationGroup folder.
2.
Set the behaviorOnInitialRegistrationReception variable with the proper behaviour.
You can also use the following line in the CLI or a configuration script:
sipEp.behaviorOnInitialRegistrationReception="Value"
where Value may be as follows:.
Table 224: Behaviour on Initial-Registration Reception Values
Value
Meaning
100
NoRegistration
200
EndpointRegistration
300
UnitRegistration
400
UnitAndEndpointRegistration
If the registration(s) succeed, then the call is re-attempted.
If the registration(s) fail, then the call is terminated.
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3.
Set the registrationDelayOnInitialRegistrationReception variable with the registration
delay, in milliseconds, on Initial-Registration Reception.
This variable configures the time interval between the unregistration confirmation (or final response)
and the registration attempt that follows.
This variable is only used when behaviorOnInitialRegistrationReception is configured to a
value other than 'NoRegistration'.
Note: This variable only applies on registration refresh triggered by the
behaviorOnInitialRegistrationReception feature.
You can also use the following line in the CLI or a configuration script:
sipEp.registrationDelayOnInitialRegistrationReception="Value"
Registration Delay Value
The quality of calls may be altered if a large quantity of registrations, more than 100, is requested at the same
time. To avoid this situation, you can configure the maximum number of seconds that the system uses to apply
a random algorithm, which is used to determine a delay before requesting a user registration or an endpoint
registration.
When the value is 0, the request registration is done immediately.
Note: The random algorithm applies individually to all registrations, meaning registrations order may not
follow their corresponding index.
 To specify the registration delay value:
1.
In the sipEpMIB, set the interopRegistrationDelayValue variable with the proper delay value.
You can also use the following line in the CLI or a configuration script:
sipEp.interopRegistrationDelayValue="Value"
where Value may be between 0 and 600 seconds.
SIP User Agent Header
This section describes configuration that is available only in the MIB parameters of the Mediatrix unit. You can
configure these parameters as follows:



by using a MIB browser
by using the CLI
by creating a configuration script containing the configuration variables
The User-Agent header field contains information about the user agent client originating the request. For
instance, the information of the User-Agent header could be something like the following:
User-Agent: Softphone Beta1.5
You can specify whether or not the Mediatrix unit sends this information when establishing a communication.
 To enable sending the SIP User Agent header:
1.
In the sipEpMIB, set the interopSendUAHeaderEnable variable to enable.
You can also use the following line in the CLI or a configuration script:
sipEp.interopSendUaHeaderEnable="1"
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Authentication Configuration
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SIP Authentication
This chapter describes how to configure authentication parameters of the Mediatrix unit.
Caution: The SIP > Authentication page is not accessible if you have the User or Observer access right.
See “Users” on page 537 for more details.
Authentication Configuration
Authentication information allows you to add some level of security to the Mediatrix unit endpoints by setting
user names and passwords.
You can add four types of authentication information:
Table 225: Authentication Information
Authentication
Description
endpoint-specific Applies only to challenges received for SIP requests related to a specific endpoint. For
instance, the registration associated with the endpoint in the user agent table or the
INVITE sent to initiate a call from the endpoint. You can define several user names and
passwords for each endpoint of the Mediatrix unit. An endpoint can thus register with
several different realms.
gateway-specific
Applies only to challenges received for SIP requests on a specific SIP gateway. You can
define several user names and passwords for each endpoint of the Mediatrix unit. An
endpoint can thus register with several different realms.
unit
Applies to all challenges received for SIP dialog. You can define several user names
and passwords for the Mediatrix unit. These user names and passwords apply to all
endpoints of the unit.
user namespecific
Applies only to challenges for a context that uses a specific user name.
The Authentication table may have between 20 and 100 rows. Each of these rows can either be associated
with the unit, a specific gateway, a specific endpoint, or a specific user name. If you have less than 20 rows,
the Mediatrix unit automatically adds new rows up to the minimum of 20.
When a challenge occurs (either 401 or 407), the first entry in the Authentication table that matches the user
name/password request is used to reply to the challenge. You can configure the use name and password in
the web interface. The order of the tried entries in the Authentication table is from the first row to the last row.
The challenge matches an authentication entry if the realm of the challenge matches the realm specified in
the Realm field or if the Validate Realm field is set to disable. For each entry matching certain criteria
(described below), the challenge is replied with the entry's user name and password. If no entry matches the
criteria, the authentication fails. To match the authentication request, the entry must also meet one of the
following criteria:
Dgw v2.0 Application

The challenge needs to be for a SIP request related to the endpoint specified in the Endpoint
column if the corresponding Apply To column is set to Endpoint.

The challenge needs to be for a SIP request performed on the SIP gateway specified in the
Gateway column if the corresponding Apply To column is set to Gateway.

The challenge needs to be for a context that uses the user name specified in the User Name
field if the corresponding Apply To column is set to Usename. The user name associated with
a context is:
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•
the user name of the FROM if the context sent the original SIP request, or
•
the user name of the request URI if the context received the original SIP request
The challenge applies to a unit if the corresponding Apply To column is set to Unit.
Creating/Editing an Authentication Entry
The web interface allows you to create authentication entries or modify the parameters of an existing one.
 To create or edit SIP authentication parameters:
1.
In the web interface, click the SIP link, then the Authentication sub-link.
Figure 117: SIP Configuration – Authentication Web Page
A
C
B
D
2.
Do one of the following:
a.
If you want to add an authentication entry before an existing entry, locate the proper row in the
table and click the
button of this row.
If you want to add an authentication entry at the end of the existing rows, click the
button
at the bottom right of the Authentication section.
b.
If you want to add several authentication entries at the same time, enter the number of entries
you want to add in the Number of rows to add at the bottom of the page.
c.
If you want to edit a single authentication entry, locate the proper row in the table and click the
button.
d.
If you want to edit several authentication entries of the current page at the same time, click the
Edit All Entries button at the bottom of the page.
This brings you to the proper Authentication panel.
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Table 226: Authentication Panel – Single Entry
3
4
5
6
7
8
9
10
Table 227: Authentication Panel – Page
3
3.
4
5
6
7
8
9
10
Select which criterion to use for matching an authentication request with an authentication entry in
the Criteria column.
Table 228: Authentication Entity
Parameter
4.
Description
Unit
The authentication entry is used on all challenges.
Endpoint
The authentication entry used for all challenges related to
a specific endpoint.
Gateway
The authentication entry is used for all challenges related
to a specific SIP gateway.
Username
The authentication entry is used for all challenges related
to a specific user name..
Enter a string that identifies an endpoint in the UserAgent.
This field is available only if you have selected Endpoint in the Criteria column for the specific row.
5.
Enter a string that identifies a SIP gateway in the GatewayStatus table.
This field is available only if you selected Gateway in the Criteria column for the specific row.
6.
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Enter a string that identifies a username in the SIP request to authenticate. this fiel is available only
if you selected Username in the Criteria column for the specific row.
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7.
Authentication Configuration
Select whether or not the current credentials are valid for any realm in the corresponding Validate
Realm drop-down menu.
Table 229: Realm Authentication Parameters
Parameter
8.
Description
Disable
The current credentials are valid for any realm. The corresponding Realm field is
read-only and cannot be modified.
Enable
The credentials are used only for a specific realm set in the corresponding Realm
field.
Enter a realm for each authentication row in the Realm column.
When authentication information is required from users, the realm identifies who requested it.
9.
Enter a string that uniquely identifies this endpoint in the realm in the User Name column.
10.
Enter a user password in the Password column.
11.
If you do not need to set other parameters, do one of the following:
•
To save your settings without refreshing the registration, click Submit.
•
To save your settings and refresh the registration now, click Submit & Refresh
Registration.
Moving an Authentication Entry
The order of the tried entries in the Authentication table is from the first row to the last row. The rows sequence
is thus very important. If you want the unit to try to match one row before another one, you must put that row
first.
 To move an authentication entry up or down:
1.
Either click the
or
arrow of the row you want to move until the entry is properly located.
Deleting an Authentication Entry
You can delete an authentication row from the table in the web interface.
 To delete an authentication entry:
1.
270
Click the
button of the row you want to delete.
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SIP Transport Type
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SIP Transport Parameters
This chapter describes the SIP transport parameters you can set.
SIP Transport Type
You can globally set the transport type for all the endpoints of the Mediatrix unit to either UDP (User Datagram
Protocol), TCP (Transmission Control Protocol), or TLS (Transport Layer Security).
The Mediatrix unit will include its supported transports in its registrations.
Please note that RFC 3261 states the implementations must be able to handle messages up to the maximum
datagram packet size. For UDP, this size is 65,535 bytes, including IP and UDP headers. However, the
maximum datagram packet size the Mediatrix unit supports for a SIP request or response is 5120 bytes
excluding the IP and UDP headers. This should be enough, as a packet is rarely bigger than 2500 bytes.
 To set the SIP transport type parameters:
1.
In the web interface, click the SIP link, then the Transport sub-link.
Figure 118: SIP Configuration – Transport Web Page
2
4
6
7
2.
8
7
8
7
3
5
8
In the General Configuration section, enable or disable the transport registration in the Add SIP
Transport in Registration drop-down menu.
When enabled, the Mediatrix unit includes its supported transports in its registrations. It registers
with one contact for each transport that is currently enabled. Each of these contacts contains a
“transport” parameter.
This is especially useful for a system where there are no SRV records configured to use a
predefined transport order for receiving requests. When sending a request, the unit either follows
the SRV configuration, or, if not available, any transport parameter received from a redirection or
from a configured SIP URL.
Note: If the Mediatrix unit has the following configuration:
• the Add SIP Transport in Registration drop-down menu is set to Disable
• the UDP transport type is disabled
• the TCP transport type is enabled
The unit will not work properly unless the SIP server uses the TCP transport type by default.
This is also true if the Mediatrix unit has the TCP transport disabled and the UDP transport enabled. In this
case, the unit will not work properly unless the SIP server uses the UDP transport protocol by default.
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3.
SIP Transport Type
Indicate whether or not the unit must include its supported transport in the Contact header in the
Add SIP Transport in Contact Header drop-down menu.
The supported transports are included in all SIP messages that have the Contact header, except
for the REGISTER message.
Available values are Enable and Disable. If you set the menu to Enable, the Mediatrix unit will send
SIP messages with the “transport” parameter in the Contact header set to:
•
transport=tcp when TCP is enabled and UDP is disabled
•
transport=udp when UDP is enabled and TCP disabled
•
no transport parameter when both TCP and UDP are enabled
•
transport=tls when secure transport (TLS) is selected
4.
Define the base port used to establish persistent connections with SIP servers when the TLS
transport is enabled in the Persistent Base Port field.
5.
Set the time interval, in seconds, before retrying the establishment of a persistent connection in the
Failback Interval field.
This is the interval that the Mediatrix unit waits before retrying periodically to establish a persistent
connection using a single IP address or a FQDN. This timer is started when a persistent connection
goes down or fails to connect to the destination.
A gateway automatically performs failover and failback procedure when it is configured with more
than one transport protocol or when responses to DNS queries return more than one IP address.
The failover and failback procedures are triggered when a SIP transaction fails. The type of
transaction that enables the failover depends on the type of the gateway. Additional SIP conditions
can be configured through SipFailoverConditions variables (see “SIP Failover Conditions” on
page 277).
During failover, the failed transaction is reattempted on another IP address with a lower priority
(according to the DNS response). When the Failback Interval expires, the IP address with a higher
priority is reattempted.
6.
In the TLS Trusted Certificate Level field, define how a peer certificate is considered trusted for a
TLS connection.
Table 230: Certificate Trust Level for TLS Connections Parameters
Parameter
7.
Description
Locally
Trusted
A certificate is considered trusted when the certificate authority (CA) that signed the
peer certificate is present in the Others Certificates table (see “Chapter 49 Certificates Management” on page 501 for more details). The certificate revocation
status is not verified.
OCSP
Optional
A certificate is considered trusted when it is locally trusted and is not revoked by its
certificate authority (CA). The certificate revocation status is queried using the
Online Certificate Status Protocol (OCSP). If the OCSP server is not available or
the verification status is unknown, the certificate is considered trusted.
OCSP
Mandatory
A certificate is considered trusted when it is locally trusted and is not revoked by its
certificate authority (CA). The certificate revocation status is queried using the
Online Certificate Status Protocol (OCSP). If the OCSP server is not available or
the verification status is unknown, the certificate is considered not trusted.
Set the TCP Connect Timeout field with the maximum time, in seconds, the unit should try to
establish a TCP connection to SIP hosts.
This timeout value is useful to have a faster detection of unreachable remote hosts. This timer can
also affect the TLS connection establishment time.
8.
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In the Protocol Configuration section, enable or disable the UDP, TCP, and TLS transport type to
use in their corresponding drop-down menu.
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Endpoint gateway supports UDP transport only. On a Trunk gateway, UDP and TCP are mutually
exclusive with TLS. Activating TLS automatically disables these unsecure protocols.
The successful configuration of a secure transport requires a little more than the activation of the
TLS protocol itself. You need to:
•
synchronize the time in the unit (see “Time Configuration” on page 58 & “SNTP
Configuration” on page 57 for more details).
•
install the security certificates used to authenticate the server to which you will connect
(see “Chapter 49 - Certificates Management” on page 501 for more details).
•
Use secure media (see “Security” on page 201 for more details).
•
configure the unit so that a “transport=tls” parameter is added to the Contact header of
your SIP requests (see Step 3).
Caution: If you have enabled Secure RTP (SRTP) on at least one line, it is acceptable to have the secure
SIP transport (TLS) disabled for testing purposes. However, you must never use this configuration in a
production environment, since an attacker could easily break it. Enabling TLS for SIP Transport is strongly
recommended and is usually mandatory for security interoperability with third-party equipment.
9.
Set the priority order of each transport type in the corresponding QValue field.
A qvalue parameter is added to each contact. The qvalue gives each transport a weight, indicating
the degree of preference for that transport. A higher value means higher preference.
The format of the qvalue string must follow the RFC 3261 ABNF (a floating point value between
0.000 and 1.000). If you specify an empty string, no qvalue is set in the contacts.
10.
Click Submit if you do not need to set other parameters.
Additional Transport Parameters
This section describes configuration that is available only in the MIB parameters of the Mediatrix unit. You can
configure these parameters as follows:



by using a MIB browser
by using the CLI
by creating a configuration script containing the configuration variables
Transport TLS Cipher Suite Settings
You can define the allowed cipher suites for the network security settings when using TLS connection.
Table 231: Cipher Suites Configuration Parameters
Parameter
CS1
Dgw v2.0 Application
Description
This is the default value and represents the cipher suites configuration prior to this variable
being introduced. This should be changed if additional network security is required. This
value includes the following cipher suites:
•
TLS_RSA_WITH_AES_256_CBC_SHA
•
TLS_RSA_WITH_AES_128_CBC_SHA
•
TLS_RSA_WITH_3DES_EDE_CBC_SHA
•
TLS_RSA_WITH_RC4_128_SHA·
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Table 231: Cipher Suites Configuration Parameters
Parameter
CS2
CS3
Description
This represents a secure configuration using SHA-1. This value includes the following cipher
suites:
•
TLS_RSA_WITH_AES_128_CBC_SHA
•
TLS_RSA_WITH_AES_256_CBC_SHA
•
TLS_RSA_WITH_3DES_EDE_CBC_SHA
•
TLS_DHE_RSA_WITH_AES_128_CBC_SHA
•
TLS_DHE_RSA_WITH_AES_256_CBC_SHA
•
TLS_DHE_RSA_WITH_3DES_EDE_CBC_SHA
This represents the most secure configuration using SHA-2. Only the most secure cipher
suites are allowed when using this value.
•
TLS_ECDHE_RSA_WITH_AES_256_GCM_SHA384
•
TLS_ECDHE_RSA_WITH_AES_256_CBC_SHA384
•
TLS_DHE_RSA_WITH_AES_256_GCM_SHA384
•
TLS_DHE_RSA_WITH_AES_256_CBC_SHA256
•
TLS_ECDHE_RSA_WITH_AES_128_CBC_SHA256
•
TLS_ECDH_RSA_WITH_AES_256_GCM_SHA384
•
TLS_ECDH_RSA_WITH_AES_256_CBC_SHA384
•
TLS_RSA_WITH_AES_256_GCM_SHA384
•
TLS_RSA_WITH_AES_256_CBC_SHA256
•
TLS_ECDHE_RSA_WITH_AES_128_GCM_SHA256
•
TLS_DHE_RSA_WITH_AES_128_GCM_SHA256
•
TLS_DHE_RSA_WITH_AES_128_CBC_SHA256
•
TLS_ECDH_RSA_WITH_AES_128_GCM_SHA256
•
TLS_CDH_RSA_WITH_AES_128_CBC_SHA256
•
TLS_RSA_WITH_AES_128_GCM_SHA256
•
TLS_RSA_WITH_AES_128_CBC_SHA256·
 To set the TLS transport cipher suite configuration parameter:
1.
In the SipEpMIB, set the TLS transport cipher suite configuration in the TransportTlsCipherSuite
variable.
You can also use the following line in the CLI or a configuration script:
SipEp.TransportTlsCipherSuite="Value"
where Value may be as follows:
Table 232: Cipher Suites Configuration Values
Value
Meaning
100
CS1
200
CS2
300
CS3
Transport Tls Version Settings
You can define the allowed TLS versions when using TLS persistent connections.
You can configure this parameter as follows:

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by using a MIB browser
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

Software Configuration Guide
by using the CLI
by creating a configuration script containing the configuration variables
Table 233: Tls Version Configuration Settings
Parameter
Description
SSLv3
Allow SSL version 3 and all TLS versions.
TLSv1
Allow TLS versions 1 and up.
TLSv1_1
Allow TLS versions 1.1 and up.
TLSv1_2
Allow TLS versions 1.2 and up.
The default value is TLSv1.
 To set the Transport Tls Version configuration parameter:
1.
In the SipEpMIB, locate the TransportGroup folder.
2.
Set theTransport Tls Version configuration in the TransportTlsVersion parameter.
You can also use the following line in the CLI or a configuration script:
SipEp.TransportTlsVersion ="Value"
where value may be:
Table 234: Tls Version Configuration Values
Value
Meaning
100
SSLv3
200
TLSv1
300
TLSv1_1
400
TLSv1_2
UDP Source Port Behaviour
On Trunk gateway type, you can configure whether or not the Mediatrix unit always uses the same local port
(the port on which it is listening for incoming packets) when sending SIP traffic over UDP. This is called
symmetric UDP source port. Symmetric UDP ports are sometimes needed to traverse NAT/Firewall devices.
When changing this setting, all destinations are automatically sent out of the penalty box, when applicable.
This variable has no effect on Endpoint gateways. Endpoint gateways always use the same UDP port for
sending and receiving messages.
The following parameters are available:
Table 235: UDP Source Port Parameters
Parameter
Dgw v2.0 Application
Description
disable
The SIP signalling over UDP uses a randomly-generated originating port. ICMP errors are
processed correctly.
enable
The SIP signalling sent over UDP originates from the same port as the port on which the user
agent is listening. ICMP messages are not processed, which means that unreachable targets
will take longer to detect.
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 To set the UDP source port behaviour:
1.
In the sipEpMIB, set whether or not the unit uses the symmetric source port feature in the
interopSymmetricUdpSourcePortEnable variable.
You can also use the following line in the CLI or a configuration script:
sipEp.interopSymmetricUdpSourcePortEnable="Value"
where Value may be as follows:
Table 236: UDP Source Port Values
Value Meaning
2.
0
disable
1
enable
Restart the SipEp service by accessing the scmMIB and setting the serviceCommandsRestart
variable for the SipEp service to restart.
You can also use the following line in the CLI or a configuration script:
scm.serviceCommands.Restart[Name=SipEp]="10"
TLS Client Authentication
When acting as a TLS server, it is customary not to request from the clients that they authenticate themselves
via the TLS protocol. However, if mutual authentication is required between client and server, you can set the
Mediatrix unit so that it requests client authentication when acting as a TLS server.
The following parameters are available:
Table 237: TLS Client Authentication Parameters
Parameter
Description
disable
The Mediatrix unit does not require TLS clients to provide their host certificate for the
connection to be allowed. This is the default value.
enable
The TLS clients must provide their host certificate for the connection to be allowed. In this case,
the level of security used to validate the host certificate is TrustedCertificate, whatever the
value set in the Certificate Validation drop-down menu of the TLS Interop section ( SIP >
Interop web page). See “TLS Interop” on page 286 for more details.
 To set TLS client authentication:
1.
In the sipEpMIB, set whether or not the Mediatrix unit requests client authentication when acting as
a TLS server in the interopTlsClientAuthenticationEnable variable.
You can also use the following line in the CLI or a configuration script:
sipEp.interopTlsClientAuthenticationEnable="Value"
where Value may be as follows:
Table 238: TLS Client Authentication Values
Value Meaning
0
disable
1
enable
Force DNS NAPTR In TLS
The Mediatrix unit allows you to force a DNS NAPTR request when the SIP transport is TLS.
This variable only applies to calls over TLS when the Supported DNS Queries drop-down menu of the SIP >
Misc page is set to NAPTR (see “DNS Configuration” on page 315 for more details).
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The following parameters are available:
Table 239: Force DNS NAPTR in TLS Parameters
Parameter
Description
disable
The DNS SRV request is sent directly with the SIP transport in SIP URI as recommended in
RFC 3263, section 4.1.
enable
A DNS NAPTR request is sent to obtain the DNS record associated with SIP over TLS. An SRV
request is performed afterward. If no SIP over TLS entry is returned, the call fails.
 To force DNS NAPTR in TLS:
1.
In the sipEpMIB, set whether or not to force a DNS NAPTR request in the
InteropForceDnsNaptrInTls variable.
You can also use the following line in the CLI or a configuration script:
sipEp.interopForceDnsNaptrInTls="Value"
where Value may be as follows:
Table 240: Force DNS NAPTR in TLS Values
Value Meaning
0
disable
1
enable
SIP Failover Conditions
You can configure additional SIP-level conditions for failover. These conditions can also be configured
specifically per gateway.
 To set the SIP failover conditions:
1.
In the sipEpMIB, set the DefaultSipFailoverConditions variable to the proper SIP failover
condition value.
You can also use the following line in the CLI or a configuration script:
sipEp.defaultSipFailoverConditions="Value"
where Value is a sequence of keywords separated by commas; spaces and tabs are ignored. If
Value is empty, only the connection-level failover conditions apply.
Supported keywords list is:
•
5xxOnRegistration: 5xx (Server Failure) response received on a registration attempt.
Note: The syntax is designed to support multiple keywords even though only a single keyword is defined
for now.
2.
If you want to set failover conditions for a specific SIP gateway, set the following variables:
•
gwSpecificFailoverEnableConfig variable for the specific SIP gateway you want to
configure to enable.
•
gwSpecificFailoverSipFailoverConditions variable for the specific SIP gateway
you want to configure to the proper value.
You can also use the following lines in the CLI or a configuration script:
sipEp.gwSpecificFailover.EnableConfig[GatewayName="default"]="5xxOnRegistration"
sipEp.gwSpecificFailover.SipFailoverConditions[GatewayName="Specific_Gateway"]="
Value"
where:
•
Dgw v2.0 Application
Specific_Gateway is the name of the SIP gateway you want to configure.
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•
Value is a SIP failover condition as defined in Step 1.
Persistent Port Interval
You can set the interval used to cycle through a range of ports.
 To set the persistent port interval:
1.
In the sipEpMIB, set the TransportPersistentPortInterval parameter or,
2.
In the CLI or a configuration script, use:
sipEp.TransportPersistentPortInterval = "Value".
Where a value equal to 0 indicates that the cycling mechanism is disabled.
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Behavior on T.38 INVITE Not Accepted
C
Software Configuration Guide
H A P T E R
31
Interop Parameters
This chapter describes the interoperability parameters that allow the Mediatrix unit to properly work,
communicate, or connect with specific IP devices.
Behavior on T.38 INVITE Not Accepted
This section describes the unit’s behaviour after receiving an error to a SIP INVITE for T.38 fax.
 To set the T.38 interop parameters:
1.
In the web interface, click the SIP link, then the Interop sub-link.
Figure 119: SIP – Interop Web Page
2
2.
In the Behavior on T.38 INVITE Not Accepted section, for each of 406, 415, 488, and 606 SIP code,
set the behaviour after receiving the code in the error response to an INVITE for T.38 fax in the
corresponding Behavior drop-down menu.
Table 241: Behavior on T.38 INVITE Not Accepted Parameters
Behavior
Description
Drop Call
The call is dropped by sending a BYE.
ReInviteForClearChannelOnly
A re-INVITE is sent with audio codecs that support clear channel
faxes.
Re-Establish Audio
A re-INVITE is sent to re-establish the audio path. Also, fax
detection is disabled for the remainder of the call.
UsePreviousMediaNegotiation No re-INVITE is sent and the audio codec from the last
successful negotiation is used. For the remainder of the call,
T.38 is disabled and fax detection may trigger a switch to a clear
channel codec that was available in the last successful
negotiation.
3.
Click Submit if you do not need to set other parameters.
SIP Interop
This section describes the SIP interoperability parameters of the Mediatrix unit .
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SIP Interop
 To set the SIP interop parameters:
1.
In the SIP Interop section of the Interop page, set whether or not the “x-Siemens-Call-Type” header
is added to the SIP packets sent by the unit in the Secure Header drop-down header.
You can set the Mediatrix unit so that it triggers the addition of the “x-Siemens-Call-Type” header to
the SIP packets sent by the unit when secure transport is in use.
The following parameters are available:
Table 242: Secure Transport Header Parameters
Parameter
Description
disable
The “x-Siemens-Call-Type” header is not added to the SIP packets sent by the
unit.
enable
The “x-Siemens-Call-Type” header is added to the SIP packets sent by the unit, and
assigned the value “ST-secure”, as soon as secure transport and secure payload
are being used. If secure transport or secure payload are not used, the header is
not added.
Figure 120: SIP Interop Section
1
2
3
5
4
6
7
8
2.
Select the username to use when the username is empty or undefined in the Default Username
Value drop-down menu.
Table 243: Default Username Value
Parameter
3.
Description
Anonymous
Sets the username to “anonymous”.
Host
Sets the username to the same value as the host.
Define the behaviour of the Mediatrix unit when answering a SIP OPTIONS request in the
OPTIONS Method Support drop-down menu.
Table 244: OPTIONS Method Support Parameters
Parameter
280
Description
None
The Mediatrix unit responds with an error 405 Method not
allowed.
AlwaysOK
The Mediatrix unit responds with a 200 OK regardless of
the content of the OPTIONS request.
Dgw v2.0 Application
SIP Interop
Software Configuration Guide
4.
Define whether or not the SIP OPTIONS requests should be ignored when all endpoints are
unusable in the Ignore OPTONS on no usable endpoints drop-down menu.
Table 245: Ignore SIP Options Parameters
Parameter
Description
Enable
The unit ignores SIP OPTIONS requests when all
endpoints are unusable. When at least one endpoint is
usable, then the SIP OPTIONS requests are answered
as configured in the OPTIONS Method Support dropdown menu (see Step 10).
Disable
The SIP OPTIONS requests are answered as configured
in the OPTIONS Method Support drop-down menu (see
Step 10) regardless of the state of the endpoints.
Note that this feature may be influenced by whether or not you have enabled the Monitor Link State
parameter. For more information:
5.
•
ISDN PRI interface: “PRI Configuration” on page 155
•
ISDN BRI interface: “BRI Configuration” on page 167
•
R2 PRI interface: “R2 Channel Associated Signaling” on page 194
Set the value of the user parameter in SIP URIs sent by the unit in the SIP URI User Parameter
Value field.
If you leave the field empty, the parameter is not added.
E.g : sip:1234@domain.com;user=InteropSipUriUserParameterValue
Note that when the Map Plus To TON International drop-down menu is set to Enable, the
parameter's value might be overwritten (“Misc Interop” on page 287).
6.
Set the Behavior On Machine Detection drop-down menu with the SIP device’s behaviour when a
machine (fax or modem) is detected during a call.
Table 246: Behavior on Machine Detection Parameters
Parameter
Description
Re-INVITE On Fax T38 Only
A SIP re-INVITE is sent only on a fax detection and T.38
is enabled.
Re-INVITE On No Negotiated Data
Codec
A SIP re-INVITE is sent on a fax or modem detection if no
data codec was previously negotiated in the original SDP
negotiation. In the case where at least one data codec
was previously negotiated in the SDP negotiation, the
device switches silently to a data codec without sending a
SIP re-INVITE. Note that if there is no data codec
enabled on the device, no SIP re-INVITE is sent and the
call is dropped by sending a BYE.
Re-INVITE Unconditional
A SIP re-INVITE is sent with data codecs upon detection
of a fax or modem even if a data codec was negotiated in
the initial offer-answer. The T.38 codec is offered if it is
enabled and a fax is detected.
See “Data Codec Selection Procedure” on page 221 for more details on the procedure the Mediatrix
unit follows when selecting data codec.
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7.
SDP Interop
Set the Registration Contact Matching field with the matching behaviour for the contact header
received in positive responses to REGISTER requests sent by the unit.
Table 247: Registration Contact Matching Parameters
Parameter
8.
Description
Strict
Matches the complete contact's SIP URI including any URI parameters, if any, as
per RFC 3261 sections '10.2.4 Refreshing Bindings' and '19.1.4 URI Comparison'.
The contact's SIP URI of a 2XX positive response MUST match the contact's SIP
URI of the REGISTER request.
Ignore Uri
Parameter
s
Matches the username and the host port part of the contact's SIP URI. All URI
parameters are ignored.
Ignore URI
and Port
Parameter
s
Matches the username part of the contact’s SIP URI. ALL URI and host port
parameters are ignored.
Set the Transmission Timeout field with the time to wait for a response or an ACK before
considering a transaction timed out.
This corresponds to timers B, F and H for all transport protocols and timer J for UDP. These timers
are defined in section A of RFC 3261.
This timeout affects the number of retransmissions. Retransmissions continue to follow the timing
guidelines described in RFC 3261.
If a DNS SRV answer contains more than one entry, the Mediatrix unit will try these entries if the
entry initially selected does not work. You can configure the maximum time, in seconds, to spend
waiting for answers to messages, from a single source. Retransmissions still follow the algorithm
proposed in RFC 3261, but the total wait time can be overridden by using this feature.
For example, if you are using DNS SRV and more than one entry are present, this timeout is the
time it takes before trying the second entry.
Available values are from 1 to 32 seconds.
9.
Click Apply if you do not need to set other parameters.
SDP Interop
This section describes the SDP interoperability parameters of the Mediatrix unit.
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 To set the SDP interop parameters:
1.
In the SDP Interop section of the Interop page, Offer Answer Model part, select the codec
negotiation rule when generating a SDP answer in the Answer Codec Negotiation drop-down menu.
Table 248: Answer Codec Negotiation Parameters
Parameter
Description
All Common - Local
Priority
When generating an answer to an offered session, all common codecs
are listed in the local order of priority. The local priority is defined for each
codec in the Telephony > CODECS page – by clicking the
button of
each codec and looking in the Voice Priority and Data Priority fields. See
“Chapter 14 - Voice & Fax Codecs Configuration” on page 181 for more
details.
First Common Local Priority
When generating an answer to an offered session, only the first common
codec with the higher local priority is listed. The local priority is defined
for each codec in the Telephony > CODECS page – by clicking the
button of each codec and looking in the Voice Priority and Data Priority
fields. See “Chapter 14 - Voice & Fax Codecs Configuration” on
page 181 for more details.
All Common - Peer
Priority
When generating an answer to an offered session, all common codecs
are listed. The codecs order is the same as in the peer offer.
First Common Peer Priority
When generating an answer to an offered session, only the first common
codec is listed. The codecs order is the same as in the peer offer.
Figure 121: SDP Interop Section
1
2
3
4
5
6
7
8
2.
Select whether or not the Mediatrix unit requires strict adherence to RFC 3264 when receiving an
answer from the peer when negotiating capabilities for the establishment of a media session in the
Enforce Offer Answer Model drop-down menu.
The following values are available:
Table 249: Offer/Answer Model Parameters
Parameter
Disable
Description
The peer can freely:
•
Send back a brand new list of codecs or add new ones to the offered
list.
•
Add new media lines.
As long as at least one codec sent back was present in the initial offer, the call
is allowed to go on. Any media line added by the peer is simply ignored.
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Table 249: Offer/Answer Model Parameters
Parameter
Enable
Description
The following guidelines from the Offer-Answer Model must be strictly
followed. An answer must:
•
Include at least one codec from the list that the Mediatrix unit sent
in the offer.
•
Contain the same number of media lines that the unit put in its offer.
Otherwise, the answer is rejected and the unit ends the call. This is the default
value.
3.
Define the behaviour of the Mediatrix unit when receiving less media announcements in the
response than in the offer in the Allow Less Media In Response drop-down menu.
The following values are available:
Table 250: Less Media Announcements Parameters
Parameter
4.
Description
Disable
The Mediatrix unit rejects the response with less media announcements than
in the offer.
Enable
The Mediatrix unit tries to find matching media when the response contains
less media announcement than in the offer. This is a deviation from the Offer/
Answer model.
Define the behaviour of the Mediatrix unit when receiving a SDP answer activating a media that had
been previously deactivated in the offer in the Allow Media Reactivation in Answer drop-down
menu.
Table 251: Media Reactivation Parameters
Parameter
5.
Description
Enable
A media reactivated in an incoming answer is ignored. This behaviour goes
against the SDP Offer/Answer model described by IETF RFC 3264.
Disable
A media reactivated in an incoming answer ends the current media negotiation
and the call. This behaviour follows the SDP Offer/Answer model described by
IETF RFC 3264.
In the Multiple Active Media part, define the behaviour of the Mediatrix unit when offering media or
answering to a media offer with audio and image negotiation in the Allow Audio and Image
Negotiation drop-down menu.
Table 252: Audio and Image Negotiation Parameters
Parameter
284
Description
Enable
The unit offers audio and image media simultaneously in
outgoing SDP offers and transits to T.38 mode upon
reception of a T.38 packet. Also, when the unit answers
positively to a SDP offer with audio and image, it transits
to T.38 mode upon reception of a T.38 packet.
Disable
Outgoing offers never include image and audio
simultaneously. Incoming offers with audio and image
media with a non-zero port are considered as offering
only audio.
Dgw v2.0 Application
SDP Interop
Software Configuration Guide
6.
Define the behaviour of the Mediatrix unit when answering a request offering more than one active
media in the Allow Multiple Active Media in Answer drop-down menu.
Figure 122: Allow Multiple Active Media in Answer
Parameter
7.
Description
disable
The answer contains only one active media. The media specified as active
in the answer is the top-most matching one in the offer. Other media are
set to inactive.
enable
Each matching active media in the offer is specified as active in the
answer. Other media are set to inactive
In the Other part, define how to set the direction attribute and the connection address in the SDP
when answering a hold offer with the direction attribute “sendonly” in the On Hold SDP Stream
Direction in Answer drop-down menu.
The following parameters are supported:
Table 253: “sendonly” Direction Attribute
Parameter
Description
inactive
The stream is marked as inactive and if the stream uses IPv4, the
connection address is set to '0.0.0.0'.
revconly
If the stream is currently active or receive only, it is marked as recvonly
and the connection address is set to the IP address of the unit.
If the stream is currently send only or inactive, it is marked as inactive
and if the stream uses IPv4, the connection address is set to '0.0.0.0'.
This method is in conformance with RFC 3264.
In both cases, no direction attribute is present in the SDP if the
interopSdpDirectionAttributeEnable variable is set to disable (see “Direction Attribute” on
page 289 for more details.
8.
Set the Codec vs Bearer Capabilities Mapping Preferred Codec Choice drop-down menu with the
behaviour of the Codec vs. Bearer Capabilities Mapping table.
This modifies the selection of the preferred codec in the incoming SDP. This parameter is available
only on ISDN interfaces.
The Codec vs. Bearer Capabilities Mapping table parameters are located in the Telephony >
CODECS > CODEC vs. Bearer Capabilities Mapping section. See “Codec vs. Bearer Capabilities
Mapping” on page 188 for more details.
Table 254: Codec vs Beareer Capabilities Mapping Preferred Codec Choice Parameters
Parameter
Dgw v2.0 Application
Description
First Codec
The first valid codec in the incoming SDP is considered
the preferred one and is used when looking up the
Codec vs. Bearer Capabilities Mapping table.
Prioritize Clear Channel
When a clear channel codec is in the incoming SDP, it
is always considered as the preferred one, no matter
where it stands in the codec list, and is used when
looking up the Codec vs. Bearer Capabilities Mapping
table.
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9.
TLS Interop
Click Submit if you do not need to set other parameters.
Note: If you are experiencing media negotiation problems (because the Mediatrix unit sends a BYE after
receiving a 200 OK), try to set the Enforce Offer Answer Model value to Disable and the Allow Less Media
In Response value to Enable.
TLS Interop
This section describes the TLS interoperability parameters of the Mediatrix unit .
 To set the TLS interop parameters:
1.
In the TLS Interop section of the Interop page, select the level of security used to validate the TLS
server certificate when the unit is acting as a TLS client in the Certificate Validation drop-down
menu.
Figure 123: TLS Interop Section
1
Note: This parameter has no effect on the TLS client authentication when the unit is acting as a TLS server
(see the interopTlsClientAuthenticationEnable variable in “TLS Client Authentication” on page 276).
The following values are available:
Table 255: TLS Certificate Validation Parameters
Parameter
2.
286
Description
No Validation
No validation of the peer certificate is performed. All TLS connections are
accepted without any verification. Note that at least one certificate must be
returned by the peer even if no validation is made. This option provides no
security and should be restricted to a lab use only.
Trusted
Certificate
Allows a TLS connection only if the peer certificate is trusted. A certificate is
considered trusted when the certificate authority (CA) that signed the peer
certificate is present in the Management > Certificates page (“Chapter 49 Certificates Management” on page 501). This option provides a minimum level
of security and should be restricted to a lab use only.
Dns Srv
Response
Allows a TLS connection if the peer certificate is trusted and contains a known
host name. A known host name can be the FQDN or IP address configured as
the SIP server, or can also be returned by a DNS SRV request. In this case,
the match is performed against the DNS response name. If it matches either
one of the Subject Alternate Name (SAN) or Common Name (CN) in the peer
certificate, the connection is allowed. This option provides an acceptable level
of security, but not as good as Host Name.
HostName
Allows a TLS connection if the peer certificate is trusted and contains a known
host name. A known host name can only be the FQDN or IP address
configured as the SIP server. If it matches either one of the Subject Alternate
Name (SAN) or Common Name (CN) in the peer certificate, the connection is
allowed. This option provides the highest level of security.
Click Submit if you do not need to set other parameters.
Dgw v2.0 Application
Misc Interop
Software Configuration Guide
Misc Interop
This section describes miscellaneous interoperability parameters of the Mediatrix unit .
 To set the Misc interop parameters:
1.
In the Misc Interop section of the Interop page, select whether or not the Mediatrix unit enables the
mapping between the “+” prefix of the user name and the “type of number” property in the Map Plus
To TON International drop-down menu.
When enabled, the service has the following behaviour:
•
For a call to SIP, the Mediatrix unit prefixes the user name with '+' if the call has the call
property “type of number” set to international. The unit also adds the “user” parameter
with the value “phone” to the SIP URI. For instance:
sip:1234@domain.com;user=phone.
•
For a call from SIP, the Mediatrix unit sets the call property “type of number” to
international if the user name has the prefix '+'.
Figure 124: Misc Interop Section
1
2
3
4
2.
Define the Ignore Plus in Username drop-down menu as to whether or not the plus (+) character is
ignored when attempting to match a challenge username with usernames in the Authentication
table.
Table 256: Ignore Plus (+) Character in Username Parameters
Parameter
3.
Description
Enable
The plus (+) character is ignored when attempting to match a username in the
authentication table.
Disable
The plus (+) character is not ignored when attempting to match a username in the
authentication table.
Select whether or not the pound character (#) must be escaped in the username part of a SIP URI
in the Escape Pound (#) in SIP URI Username drop-down menu.
Table 257: Escape Pound Parameters
Parameter
Description
Enable
The Pound character (#) is escaped in the username
part of a SIP URI.
Disable
The Pound character (#) is not escaped in the
username part of a SIP URI.
Note that RFC 3261 specifies that the pound character
(#) needs to be escaped in the username part of a SIP
URI.
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Additional Interop Parameters
Select the format of the escaped characters to be used in all SIP headers in the Escape Format
drop-down menu.
Table 258: Escape Format Parameters
Parameter
5.
Description
Lower Hexadecimal
Escaped characters are displayed in a lowercase
hexadecimals format.
Upper Hexadecimal
Escaped characters are displayed in a uppercase
hexadecimals format.
Click Submit if you do not need to set other parameters.
Additional Interop Parameters
This section describes configuration that is available only in the MIB parameters of the Mediatrix unit. You can
configure these parameters as follows:



by using a MIB browser
by using the CLI
by creating a configuration script containing the configuration variables
The interop parameters allow the Mediatrix unit to properly work, communicate, or connect with specific IP
devices.
Call Waiting Private Number Criteria for SIP INFO
You can specify the call waiting criteria, in the form of a regular expression, that defines a private number
received in a SIP INFO.
 To set the Call Waiting Private Number Criteria:
1.
In the sipEpMIB, set the Call Waiting Private Number Criteria in the
InteropCallWaitingSipInfoPrivateNumberCriteria variable.
You can also use the following line in the CLI or a configuration script:
sipEp.interopCallWaitingSipInfoPrivateNumberCriteria="Value"
For example, the value "(Anonymous|anonymous)" would define a calling number that is either
"Anonymous" or "anonymous" as private. The regular expression symbols to match the beginning
and end of the number are implicit and do not need to be specified. See “Regular Expressions” on
page 432 for more details.
The variable is effective only if the Default Hook-Flash Processing parameter of the SIP > Misc page
is set to TransmitUsingSignalingProtocol (see “General Configuration” on page 385 for more
details).
Max-Forwards Header
Max-Forwards serves to limit the number of hops a request can make on the way to its destination. It consists
of an integer that is decremented by one at each hop. If the Max-Forwards value reaches 0 before the request
reaches its destination, it is rejected with a “483 (Too Many Hops)” error response. The Max-Forwards SIP
header is always present and the default value is 70.
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Direction Attributes in a Media Stream
The Mediatrix unit allows you to define various direction attributes pertaining to the media stream.
When Putting a Call on Hold
The Mediatrix unit can provide the direction attribute and the meaning of the connection address “0.0.0.0” sent
in the SDP when an endpoint is put on hold.
The following parameters are supported:
Table 259: Direction Attributes
Parameter
Description
inactive
The stream is put on hold by marking it as inactive. This is the default value. This setting
should be used for backward compatibility issues.
sendonly
The stream is put on hold by marking it as sendonly. This method allows the Mediatrix unit to
be in conformance with RFC 3264.
 To define the direction attribute when putting a call on hold:
1.
In the sipEpMIB, set the interopOnHoldSdpStreamDirection variable to the proper value.
You can also use the following line in the CLI or a configuration script:
sipEp.interopOnHoldSdpStreamDirection="Value"
where Value may be as follows:
Table 260: Direction Attributes Values
Value
Meaning
100
inactive
200
sendonly
This configuration has no effect if the interopSdpDirectionAttributeEnable variable is set to
disable (see “Direction Attribute” on page 289 for more details).
Direction Attribute
You can define if the SDP direction attribute is supported by the unit.
This variable applies only when the negotiated media uses an IPv4 address. The application always behaves
as if this variable is set to Enable for media using an IPv6 address.
The following parameters are supported:
Table 261: SDP Direction Attribute
Parameter
disable
Description
No direction attribute is present in the SDP sent by the Mediatrix unit.
The Mediatrix unit ignores any direction attribute found in the SDP received from the
peer.
The method to put a session on hold is in conformance with RFC 2543.
enable
The Mediatrix unit always sends the direction attribute in the SDP of an initiated call. For
all other SDP messages sent by the unit, refer to “Enable/Disable SDP Detect Peer
Direction Attribute Support” on page 290.
If present in the SDP, the direction attribute is preferred over the connection address to
transmit session modification information.
This method is in conformance with RFC 3264.
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 To define if the direction attribute is present:
1.
In the sipEpMIB, set the interopSdpDirectionAttributeEnable variable to the proper value.
You can also use the following line in the CLI or a configuration script:
sipEp.interopSdpDirectionAttributeEnable="Value"
where Value may be as follows:
Table 262: SDP Direction Attribute
Value Meaning
0
disable
1
enable
Enable/Disable SDP Detect Peer Direction Attribute Support
You can define if the SDP direction attribute support should be autodetected in the SDP received from the
peer.
This variable is used only when the negotiated media uses an IPv4 address and when the
interopSdpDirectionAttributeEnable is enabled (see “Direction Attribute” on page 289 for more details).
The application always behaves as if this variable is set to 'Disable' for media using an IPv6 address.
The following parameters are supported:
Table 263: SDP Detect Peer Direction Attribute Parameters
Parameter
Description
disable
The Mediatrix unit always sends the direction attribute in the SDP without autodetection
of peer support.
enable
The initial handshake determines if the peer supports the direction attribute. The
direction attribute will be present when the peer supports it.
 To define if the SDP detect peer direction attribute is enabled or disabled:
1.
In the sipEpMIB, set the interopSdpDetectPeerDirectionAttributeSupportEnable variable to
the proper value.
You can also use the following line in the CLI or a configuration script:
sipEp.interopSdpDetectPeerDirectionAttributeSupportEnable="Value"
where Value may be as follows:
Table 264: SDP Detect Peer Direction Attribute Values
Value Meaning
0
disable
1
enable
On Hold SDP Connection Address
You can define the value of the connection address sent in the SDP when an endpoint is on hold and no longer
listening to media packets.
This variable is used only when the negotiated media uses an IPv4 address. The application always behaves
as if this variable is set to 'MediaAddress' for media using an IPv6 address.
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The following parameters are supported:
Table 265: On Hold SDP Connection Address Parameters
Parameter
Description
HoldAddress
The connection address sent in the SDP is '0.0.0.0' if the media uses an IPv4 address.
This method is described by RFC 2543.
MediaAddress
The connection address sent in the SDP is the listening address.
 To define the on hold SDP connection address:
1.
In the sipEpMIB, set the interopOnHoldSdpConnectionAddress variable to the proper value.
You can also use the following line in the CLI or a configuration script:
sipEp.interopOnHoldSdpConnectionAddress="Value"
where Value may be as follows:
Table 266: On Hold SDP Connection Address Values
Value
Meaning
100
HoldAddress
200
MediaAddress
Answering a Hold Offer with the Direction Attribute “sendonly”
You can define how to set the direction attribute in the SDP when answering a hold offer with the direction
attribute 'sendonly'.
The following parameters are supported:
Table 267: “sendonly” Direction Attribute
Parameter
Description
inactive
The stream is marked as inactive and if the stream uses an IPv4
address, the connection address is set according to the
InteropOnHoldSdpConnectionAddress variable (“On Hold SDP
Connection Address” on page 290).
revconly
If the stream is currently active or receive only, it is marked as recvonly
and the connection address is set to the IP address of the unit.
If the stream is currently send only or inactive, it is marked as inactive
and the connection address is set according to the
InteropOnHoldSdpConnectionAddress variable (“On Hold SDP
Connection Address” on page 290).
This method is in conformance with RFC 3264.
 To define the behaviour with the “sendonly” direction attribute:
1.
In the sipEpMIB, set the InteropOnHoldAnswerSdpStreamDirection variable to the proper value.
You can also use the following line in the CLI or a configuration script:
sipEp.interopOnHoldAnswerSdpStreamDirection="Value"
where Value may be as follows:
Table 268: “sendonly” Direction Attribute
Value Meaning
100
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Table 268: “sendonly” Direction Attribute (Continued)
Value Meaning
200
Recvonly
In both cases, no direction attribute is present in the SDP if the
interopSdpDirectionAttributeEnable variable is set to disable (see “Direction Attribute” on
page 289 for more details.
SDP Direction Attribute Level
You can define the preferred location where the stream direction attribute is set.
The following parameters are supported:
Table 269: SDP Direction Attribute Level
Parameter
MediaOrSessionLevel
Description
If every media have the same direction, the stream direction attribute is
only present at session level.
Otherwise, the stream direction attribute is only present at media level.
MediaAndSessionLevel
If every media have the same direction, the stream direction attribute is
present both at session level and media level.
Otherwise, the stream direction attribute is only present at media level.
 To define the SDP direction attribute level:
1.
In the sipEpMIB, set the InteropSdpDirectionAttributeLevel variable to the proper value.
You can also use the following line in the CLI or a configuration script:
sipEp.InteropSdpDirectionAttributeLevel="Value"
where Value may be as follows:
Table 270: SDP Direction Attribute Level
Value
Meaning
100
MediaOrSessionLevel
200
MediaAndSessionLevel
Local Ring Behaviour on Provisional Response
You can set the Mediatrix unit so that it starts or not the local ring upon receiving a “18x Provisional” response
without SDP.
This setting does not affect the behaviour when the “18x Provisional” response contains SDP, which allows
establishing an early media session before the call is answered.
This variable does not affect the behaviour in case the '18x Provisional' response contains SDP, in which case
the media stream, if present, is played.
The following parameters are supported:
Figure 125: Local Ring Behaviour
Parameter
Disable
292
Description
The local ring is not started on a '18x Provisional' response without SDP, with one
exception: the '180 Ringing' without SDP will start the local ring if the media stream is
not already established.
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Figure 125: Local Ring Behaviour (Continued)
Parameter
Description
LocalRingWhenNo : The local ring is started on any '18x Provisional' response without SDP if the media
EstablishedMediaS stream is not already established.
tream
LocalRingAlways
The local ring is always started on any '18x Provisional' response without SDP.
 To define the local ring behaviour on provisional response:
1.
In the sipEpMIB, set the interopLocalRingOnProvisionalResponse variable to the proper value.
You can also use the following line in the CLI or a configuration script:
sipEp.interopLocalRingOnProvisionalResponse="Value"
where Value may be as follows:
Figure 126: Local Ring Values
Value
Meaning
0
disable
1
LocalRingWhenNoEstablishedMediaStream
2
LocalRingAlways
Session ID and Session Version Number in the Origin Field of the SDP
You can define the maximum length of the session ID and the session version number in the origin line (o=)
of the SDP. This allows the Mediatrix unit to be compatible with 3rd party vendor equipment.
The following parameters are supported:
Table 271: Maximum Length Parameters
Length
Description
max-32bits
The session ID and the session version number are represented with a 32 bit integer.
They have a maximum length of 10 digits.
max-64bits
The session ID and the session version number are represented with a 64 bit integer.
They have a maximum length of 20 digits. This is the default value.
 To set the maximum length of the session ID and the session version number:
1.
In the sipEpMIB, set the interopSdpOriginLineSessionIDAndVersionMaxLength variable with
the proper length.
You can also use the following line in the CLI or a configuration script:
sipEp.interopSdpOriginLineSessionIdAndVersionMaxLength="Value"
where Value may be as follows:
Table 272: Maximum Length Values
Value
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Meaning
100
max-32bits
200
max-64bits
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Register Home Domain Override
By default, the address-of-record in the “To” header uses the value set in the Proxy Host field of the SIP/
Configuration page for the host/port part. See “SIP Servers Configuration” on page 248 for more details. You
can override this value if required.
 To override the register home domain value:
1.
In the sipEpMIB, set the interopRegisterHomeDomainOverride variable with the override home
domain value.
You can also use the following line in the CLI or a configuration script:
sipEp.interopRegisterHomeDomainOverride="IP_Address"
The address of record in the register will use this string instead of the home domain proxy. If the
variable is empty, the value of the Proxy Host field is used.
The host is also overridden in the From and Call-Id headers since they match the To header.
DNS SRV Record Lock
You can configure the Mediatrix unit to always use the same DNS SRV record for a SIP call ID. As a result, a
call or registration always uses the same destination until the destination is unreachable or the unit receives
a different DNS SRV result.
The following parameters are supported:
Table 273: DNS SRV Record Lock Parameters
Length
Description
disable
The behaviour follows RFC 3263.
enable
All messages during a call or registration use the same SRV record.
 To enable the DNS SRV record lock feature:
1.
In the sipEpMIB, set the interopLockDnsSrvRecordPerCallEnable variable to enable.
You can also use the following line in the CLI or a configuration script:
sipEp.interopLockDnsSrvRecordPerCallEnable="Value"
where Value may be as follows:
Figure 127: DNS SRV Record Lock Values
Value Meaning
0
disable
1
enable
Listening for Early RTP
Note: This feature applies to the following models:
• Mediatrix 3208 / 3216
• Mediatrix 3308 / 3316
• Mediatrix 3716 / 3731 / 3732 / 3741 / 3742
• Mediatrix 4100 Series
• Mediatrix LP Series
• Mediatrix C7 Series
You can set the Mediatrix unit so that it listens for RTP before the reception of a response with SDP. This
feature only applies to calls initiated from analog endpoints (FXS/FXO) with non-secure RTP.
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The following parameters are supported:
Table 274: Early RTP Parameters
Length
Description
enable
The RTP port is opened after the initial INVITE has been sent, without waiting for a provisional or
final response with SDP to be received. No local ring is generated. This conforms to section 5.1 of
RFC 3264.
disable
The RTP port is opened only after a response with SDP is received.
Warning: Do not enable this feature unless the server supports early RTP (or early media). Failing so
prevents any ringing to be heard for outgoing calls.
 To enable the Early RTP feature:
1.
In the sipEpMIB, set the InteropListenForEarlyRtpEnable variable to enable.
You can also use the following line in the CLI or a configuration script:
sipEp.InteropListenForEarlyRtpEnable="Value"
where Value may be as follows:
Figure 128: Early RTP Values
Value Meaning
0
disable
1
enable
Resolve Route Header
The Mediatrix unit has a parameter that allows you to resolve the FQDN in the top-most route header of
outgoing packets.
The following parameters are supported:
Table 275: Resolve Route Header Parameters
Length
Description
enable
The FQDN in the top-most route header is replaced by the IP address of the packet's destination
if the FQDN matches the gateway's configured outbound proxy.
disable
The route header is not modified.
 To resolve the route header:
1.
In the sipEpMIB, set the InteropResolveRouteHeaderEnable variable with the proper value.
You can also use the following line in the CLI or a configuration script:
sipEp.interopResolveRouteHeaderEnable="Value"
where Value may be as follows:
Figure 129: Resolve Route Header Values
Value Meaning
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0
disable
1
enable
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ACK Branch Matching
You can configure the method used to match incoming ACK SIP packets.
The following parameters are supported:
Table 276: ACK Branch Matching Parameters
Parameter
Rfc3261
Description
Follows the method described in RFC 3261 (section 8.1.1.7). The branch value in the topmost
via of the ACK request to a 2XX response MUST be different than the one of the INVITE.
Rfc3261Wi Follows the method described in RFC 3261 (section 8.1.1.7) but enables the handling of ACK
thoutAck
requests (for 2XX responses) that have the same branch value in the topmost via as the
INVITE.
 To set ACK branch matching:
1.
In the sipEpMIB, set the interopAckBranchMatching variable with the proper value.
You can also use the following line in the CLI or a configuration script:
sipEp.interopAckBranchMatching="Value"
where Value may be as follows:
Figure 130: ACK Branch Matching Values
Value
Meaning
100
Rfc3261
200
Rfc3261WithoutAck
Ignore Require Header
You can define whether or not the Require Header must be ignored when processing the incoming SIP Client
requests (INVITE, re-INVITE, Bye, etc.).
The following parameters are supported:
Table 277: Ignore Require Header Parameters
Parameter
Description
Enable
The Require Header is ignored and no validation about these options-tags is performed.
Disable
The Require Header options-tags are validated and, when an option-tag is not supported, a
420 (Bad Extension) response is sent.
The supported options-tags are:
•
* 100rel
•
* replaces
•
* timer
 To set whether or not to ignore the Require header:
1.
In the sipEpMIB, set the interopIgnoreRequireHeaderEnable variable with the proper value.
You can also use the following line in the CLI or a configuration script:
sipEp.interopIgnoreRequireHeaderEnable="Value"
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where Value may be as follows:
Figure 131: Ignore Require Header Values
Value Meaning
0
disable
1
enable
Reject Code for Unsupported SDP Offer
You can define the rejection code used when an offer is received with invalid or unsupported SDP Offer. RFC
3261 recommends using the error code 488 'Not Acceptable Here'.
The following parameters are supported:
Table 278: Reject Code for Unsupported SDP Offer Parameters
Parameter
Description
UnsupportedMediaType The SIP error code 415 'Unsupported Media Type' is returned if the ContentType is invalid; the payload is missing or the SDP content is invalid.
NotAcceptableHere
The SIP error code 488 'Not Acceptable Here' is returned if the SDP content is
invalid.
 To set the reject code:
1.
In the sipEpMIB, set the InteropRejectCodeForUnsupportedSdpOffer variable with the proper
value.
You can also use the following line in the CLI or a configuration script:
sipEp.InteropRejectCodeForUnsupportedSdpOffer="Value"
where Value may be as follows:
Figure 132: Reject Code Values
Value
Meaning
415
UnsupportedMediaType
488
NotAcceptableHere
SIP User-Agent Header Format
You can define the text to display in the SIP User-Agent header. You can use macros to include information
specific to the unit.
You can also define the same information in the HTTP User-Agent header. See“HTTP User-Agent Header
Format” on page 16 for more details.
 To set the SIP User-Agent header format:
1.
In the sipEpMIB, set the User-Agent header format in the interopUaHeaderFormat variable.
You can also use the following line in the CLI or a configuration script:
sipEp.interopUaHeaderFormat="Value"
where Value may contain any text, as well as one or more of the following macros:
Table 279: Macros Supported
Macro
%version%
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Description
Application version.
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Table 279: Macros Supported (Continued)
Macro
%mac%
Description
MAC address.
%product% Product name.
%profile%
Profile.
%%
Insert the % character.
For instance, the default value is:
%product%/v%version% %profile%
SIP INFO Without Content Answer
You can define the response of the Mediatrix unit to a received SIP INFO with no message body for an existing
call.
RFC 2976 recommends that a 200 OK response MUST be sent for an INFO request with no message body if
the INFO request was successfully received for an existing call.
The following parameters are supported:
Table 280: Reject Code for Unsupported SDP Offer Parameters
Parameter
Description
UnsupportedMediaType The unit responds with the SIP error code 415 'Unsupported Media Type'.
Ok
The unit responds with a 200 OK.
 To define the SIP INFO Without Content Answer behaviour:
1.
In the sipEpMIB, set the interopSipInfoWithoutContentAnswer variable with the proper
behaviour.
You can also use the following line in the CLI or a configuration script:
sipEp.interopSipInfoWithoutContentAnswer="Value"
where Value may be as follows:
Table 281: SIP INFO Values
Value
Meaning
200
Ok
415
UnsupportedMediaType
Keep Alive Option Format
You can configure the Keep Alive OPTION requests format.
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The following parameters are supported:
Table 282: Keep Alive Option Format Parameters
Parameter
Description
ShortFrom
The unit sends the OPTION request with the standard format with only the unit’s
IP address in the from header. This is the default.
FullFrom
The unit sends the OPTION request with the standard format with the first
registered username and IP address in the from header.
Note: The SipEp service must be restarted to apply a new username to the Keep Alive.
 To set the keep alive option format:
1.
In the sipEpMIB, locate the InteropGroup folder.
2.
Set the InteropKeepAliveOptionFormat variable with the proper value.
You can also use the following line in the CLI or a configuration script:
sipEp.InteropKeepAliveOptionFormat="Value"
where Value may be as follows:
Figure 133: Keep Alive Option Format Values
Value
Meaning
100
ShortFrom
200
FullFrom
Unsupported Content-Type
You can define the behaviour of the Mediatrix unit upon reception of a SIP packet containing multiple
unsupported Content-Type in the payload.
The following parameters are supported:
Table 283: Unsupported Content-Type Parameters
Parameter
Description
Reject
Unsupported Content-Type are rejected.
Allow
Unsupported Content-Type are allowed and ignored if at least one Content-Type
is supported.
Ignore
Unsupported Content-Type are ignored.
Note: When ignored, unsupported Content-Type are treated as if they were not present in the packet.
 To define the unsupported Content-Type behaviour:
1.
In the sipEpMIB, set the interopUnsupportedContentType variable with the proper behaviour.
You can also use the following line in the CLI or a configuration script:
sipEp.interopUnsupportedContentType="Value"
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where Value may be as follows:
Table 284: Unsupported Content-Type Values
Value Meaning
300
100
Reject
200
Allow
300
Ignore
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H A P T E R
32
Miscellaneous SIP Parameters
This chapter describes miscellaneous SIP parameters you can set:











SIP penalty box parameters
How to override the default mapping of error causes defined in RFC 3398.
Additional Headers
PRACK
Session Refresh
SIP Gateway Configuration
SIP Blind Transfer Method
Diversion Configuration
DNS Configuration
Event Handling Configuration
Messaging Subscription
SIP Penalty Box
The penalty box feature is used when a given host FQDN resolves to a non-responding address. When the
address times out, it is put into the penalty box for a given amount of time. During that time, the address in
question is considered as “non-responding” for all requests.
This feature is most useful when using DNS requests returning multiple or varying server addresses. It makes
sure that, when a host is down, users wait a minimal amount of time before trying a secondary host.
When enabled, this feature takes effect immediately on the next call attempt.
The penalty box feature is applied only when using UDP or TCP connections established with a FQDN. A
similar penalty box feature for the TLS persistent connections is available via the TLS Persistent Retry Interval
parameter. See “SIP Transport Type” on page 271 for more details.
Note: The Penality Box feature is disabled when a gateway of type "endpoint" is configured in the gateway.
Penalty Box vs Transport Types
Media5 recommends to use this feature with care when supporting multiple transports (see “Chapter 30 - SIP
Transport Parameters” on page 271 for more details) or you may experience unwanted behaviours.
When the Mediatrix unit must send a packet, it retrieves the destination from the packet. If the destination
address does not specify a transport to use and does not have a DNS SRV entry that configures which
transport to use, then the Mediatrix unit tries all transports it supports, starting with UDP. If this fails, it tries
with TCP. The unit begins with UDP because all SIP implementations must support this transport, while the
mandatory support of TCP was only introduced in RFC 3261.
Note: It is not the destination itself that is placed in the penalty box, but the combination of address, port
and transport. When a host is in the penalty box, it is never used to try to connect to a remote host unless it
is the last choice for the Mediatrix unit and there are no more options to try after this host.
Let’s say for instance that the Mediatrix unit supports both the UDP and TCP transports. It tries to reach
endpoint “B” for which the destination address does not specify a transport and there is no DNS SRV entry to
specify which transports to use in which order. It turns out that this endpoint “B” is also down. In this case, the
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Error Mapping
Mediatrix unit first tries to contact endpoint “B” via UDP. After a timeout period, UDP is placed in the penalty
box and the unit then tries to contact endpoint “B” via TCP. This fails as well and TCP is also placed in the
penalty box.
Now, let’s assume endpoint “B” comes back to life and the Mediatrix unit tries again to contact it before UDP
and TCP are released from the penalty box. First, the unit tries UDP, but it is currently in the penalty box and
there is another transport left to try. The Mediatrix unit skips over UDP and tries the next target, which is TCP.
Again, TCP is still in the penalty box, but this time, it is the last target the Mediatrix unit can try, so penalty box
or not, TCP is used all the same to try to contact endpoint “B”.
There is a problem if endpoint “B” only supports UDP (RFC 2543-based implementation). Endpoint “B” is up,
but the Mediatrix unit still cannot contact it: with UDP and TCP in the penalty box, the unit only tries to contact
endpoint “B” via its last choice, which is TCP.
The same scenario would not have any problem if the penalty box feature was disabled. Another option is to
disable TCP in the Mediatrix unit, which makes UDP the only possible choice for the unit and forces to use
UDP even if it is in the penalty box.
You must fully understand the above problem before configuring this feature. Mixing endpoints that do not
support the same set of transports with this feature enabled can lead to the above problems, so it is suggested
to either properly configure SRV records for the hosts that can be reached or be sure that all hosts on the
network support the same transport set before enabling this feature.
Penalty Box Configuration
The following steps describe how to configure the penalty box feature.
 To set the SIP penalty box parameters:
1.
In the web interface, click the SIP link, then the Misc sub-link.
Figure 134: SIP Configuration – Misc Web Page
2
3
2.
In the Penalty Box section, enable the SIP penalty box feature by selecting Enable in the Penalty
Box Activation drop-down menu.
The penalty box is always “active”. This means that even if the feature is disabled, IP addresses are
marked as invalid, but they are still tried. This has the advantage that when the feature is enabled,
IP addresses that were already marked as invalid are instantly put into the penalty box.
3.
Set the amount of time, in seconds, that a host spends in the penalty box in the Penalty Box Time
field.
Changing the value does not affect IP addresses that are already in the penalty box. It only affects
new entries in the penalty box.
4.
Click Submit if you do not need to set other parameters.
Error Mapping
You can override the default mapping of error causes defined in RFC 3398.The web interface offers two
sections:

302
The SIP To Cause Error Mapping section allows you to override the default mapping for SIP
code to ISDN cause.
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Error Mapping
Software Configuration Guide

The Cause To SIP Error Mapping section allows you to override the default mapping for ISDN
cause to SIP code.
The following standard SIP codes are available:
400: Bad Request
414: Request-URI too long
485: Ambiguous
401: Unauthorized
415: Unsupported media type
486: Busy here
402: Payment required
416: Unsupported URI Scheme
500: Server internal error
403: Forbidden
420: Bad extension
501: Not implemented
404: Not found
421: Extension Required
502: Bad gateway
405: Method not allowed
423: Interval Too Brief
503: Service unavailable
406: Not acceptable
480: Temporarily unavailable
504: Server time-out
407: Proxy authentication
required
481: Call/Transaction Does not
Exist
504: Version Not Supported
408: Request timeout
482: Loop Detected
600: Busy everywhere
410: Gone
483: Too many hops
603: Decline
413: Request Entity too long
484: Address incomplete
604: Does not exist anywhere
513: Message Too Large
You can also map any other custom code between 400 and 699.
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Error Mapping
The following standard ISDN cause numbers specified in Q.931 are available:
Normal event:
Service or option not implemented:
1: Unassigned (unallocated) number.
65: Bearer capability not implemented.
2: No route to specified transit network.
66: Channel type not implemented.
3: No route to destination.
69: Requested facility not implemented.
6: Channel unacceptable.
70: Only restricted digital information bearer.
7: Call awarded and being delivered in an
established channel.
79: Service or option not implemented,
unspecified.
17: User busy.
18: No user responding.
Invalid Message
19: User alerting, no answer.
81: Invalid call reference value.
20: Subscriber absent.
82: Identified channel does not exist.
21: Call rejected.
22: Number changed.
83: A suspended call exists, but this call identity
does not.
23: Redirection to new destination.
84: Call identity in use.
26: Non-selected user clearing.
85: No call suspended.
27: Destination out of order.
86: Call having the requested call identity has been
cleared.
28: Invalid number format (incomplete number).
29: Facility rejected.
30: Response to STATUS ENQUIRY.
31: Normal, unspecified.
Resource unavailable:
34: No circuit/channel available.
38: Network out of order.
41: Temporary failure.
42: Switching equipment congestion.
43: Access information discarded.
44: Requested circuit/channel not available.
47: Resource unavailable, unspecified.
87: user not member of CUG.
88: Incompatible destination.
91: Invalid transit network selection.
95: Invalid message, unspecified.
Protocol error
96: Mandatory information element is missing.
97: Message type non-existent or not
implemented.
98: Message not compatible with call state or
message type non-existent or not implemented.
99: Information element non-existent or not
implemented.
100: Invalid information element contents.
Service or option not available:
101: Message not compatible with call state.
55: Incoming calls barred within CUG.
102: Recovery on time expiry.
57: Bearer capability not authorized.
111: Protocol error, unspecified.
58: Bearer capability not presently available.
Interworking
63: Service or option not available, unspecified.
127: Interworking, unspecified
You can also map any other custom code between 1 and 127.
SIP to Cause Error Mapping
This section describes how to override the default mapping of ISDN error causes.
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 To override the default mapping of ISDN error causes:
1.
In the SIP To Cause Error Mapping section of the Misc page, click the
button to add a new row.
Figure 135: SIP To Cause Error Mapping Section
1
This brings you to the Configure New SIP To Cause Error Mapping panel.
2.
Enter the SIP code in the SIP Code field, then the corresponding ISDN cause number in the Cause
column.
You can use the Suggestion column’s drop-down menu to select between available code values.
Figure 136: Configure New SIP To Cause Error Mapping Panel
2
3.
Click Submit.
This brings you back to the main Misc web page.
You can delete an existing row by clicking the
button.
You can modify the Cause value by typing a new code in the field. See “SIP To Cause Default Error
Mapping” on page 305 for the default mappings as per RFC 3398.
4.
Click Submit if you do not need to set other parameters.
SIP To Cause Default Error Mapping
Table 285 lists the default mappings as per RFC 3398.
Table 285: SIP To Cause Default Error Mapping
SIP Response Received
Dgw v2.0 Application
Cause Value
400 Bad Request
41
Temporary Failure
401 Unauthorized
21
Call rejected
402 Payment required
21
Call rejected
403 Forbidden
21
Call rejected
404 Not found
1
Unallocated number
405 Method not allowed
63
Service or option unavailable
406 Not acceptable
79
Service/option not implemented
407 Proxy authentication required
21
Call rejected
408 Request timeout
102 Recovery on timer expiry
410 Gone
22
413 Request Entity too long
127 Interworking
414 Request-URI too long
127 Interworking
Number changed (w/o diagnostic)
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Error Mapping
Table 285: SIP To Cause Default Error Mapping (Continued)
SIP Response Received
Cause Value
415 Unsupported media type
79
Service/option not implemented
416 Unsupported URI Scheme
127 Interworking
420 Bad extension
127 Interworking
421 Extension Required
127 Interworking
423 Interval Too Brief
127 Interworking
480 Temporarily unavailable
18
No user responding
481 Call/Transaction Does not Exist
41
Temporary Failure
482 Loop Detected
25
Exchange - routing error
483 Too many hops
25
Exchange - routing error
484 Address incomplete
28
Invalid Number Format
485 Ambiguous
1
Unallocated number
486 Busy here
17
User busy
500 Server internal error
41
Temporary failure
501 Not implemented
79
Not implemented, unspecified
502 Bad gateway
38
Network out of order
503 Service unavailable
41
Temporary failure
504 Server time-out
102 Recovery on timer expiry
504 Version Not Supported
127 Interworking
513 Message Too Large
127 Interworking
600 Busy everywhere
17
User busy
603 Decline
21
Call rejected
604 Does not exist anywhere
1
Unallocated number
Cause to SIP Error Mapping
This section describes how to override the default mapping of SIP codes.
 To override the default mapping of SIP codes:
1.
In the Cause To SIP Error Mapping section of the Misc page, click the
button to add a new row.
Figure 137: Cause To SIP Error Mapping Section
1
This brings you to the Configure New Cause To SIP Error Mapping panel.
2.
Enter the ISDN cause number in the Cause column, then the corresponding SIP code in the SIP
Code field.
You can use the Suggestion column’s drop-down menu to select between available code values.
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Error Mapping
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Figure 138: Configure New Cause To SIP Error Mapping Panel
2
3.
Click Submit.
This brings you back to the main Misc web page.
You can delete an existing row by clicking the
button.
You can modify the SIP Code value by typing a new code in the field. See “Cause To SIP Default
Error Mapping” on page 307 for the default mappings as per RFC 3398.
4.
Click Submit if you do not need to set other parameters.
Cause To SIP Default Error Mapping
Table 286 lists the default mappings as per RFC 3398.
Table 286: Cause To SIP Default Error Mapping
ISUP Cause Value
SIP Response
Normal Event
1
unallocated number
404 Not Found
2
no route to network
404 Not Found
3
no route to destination
404 Not Found
16
normal call clearing
---
17
user busy
486 Busy Here
18
no user responding
408 Request Timeout
19
no answer from the user
480 Temporarily unavailable
20
subscriber absent
480 Temporarily unavailable
21
call rejected
403 Forbidden
22
number changed (w/o diagnostic)
410 Gone
22
number changed (w/ diagnostic)
301 Moved Permanently
23
redirection to new destination
410 Gone
26
non-selected user clearing
404 Not Found
27
destination out of order
502 Bad Gateway
28
address incomplete
484 Address incomplete
29
facility rejected
501 Not implemented
31
normal unspecified
480 Temporarily unavailable
BYE or CANCEL
Resource Unavailable
Dgw v2.0 Application
34
no circuit available
503 Service unavailable
38
network out of order
503 Service unavailable
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Chapter 32 - Miscellaneous SIP Parameters
Error Mapping
Table 286: Cause To SIP Default Error Mapping (Continued)
ISUP Cause Value
SIP Response
41
temporary failure
503 Service unavailable
42
switching equipment congestion
503 Service unavailable
47
resource unavailable
503 Service unavailable
Service or Option not Available
55
incoming calls barred within CUG
403 Forbidden
57
bearer capability not authorized
403 Forbidden
58
bearer capability not presently available 503 Service unavailable
Service or Option not Implemented
65
bearer capability not implemented
488 Not Acceptable Here
70
only restricted digital available
488 Not Acceptable Here
79
service or option not implemented
501 Not implemented
Invalid message
87
user not member of CUG
403 Forbidden
88
incompatible destination
503 Service unavailable
Protocol error
102 recovery of timer expiry
504 Gateway timeout
111
500 Server internal error
protocol error
Interworking
127 interworking unspecified
308
500 Server internal error
Dgw v2.0 Application
Additional Headers
Software Configuration Guide
Additional Headers
You can define whether or not the Mediatrix unit uses additional SIP headers.
 To use additional SIP headers:
1.
In the Additional Headers section of the Misc page, select the method to use in the Reason Header
Support drop-down menu.
Figure 139: Reason Header Section
1
2
Table 287: Reason Header Support Parameters
Parameter
2.
Description
None
Silently ignores any incoming reason headers and does not
send the reason header.
SendQ850
Silently ignores incoming reason codes and sends the SIP
reason code when the original Q.850 code is available. The
reason code sent is not affected by the entries in the Error
Mapping SIP To Cause table.
ReceiveQ850
Uses the incoming Q.850 reason cause header. When
received, the reason code supersedes any entrie s in the
Error Mapping SIP To Cause table.
SendReceiveQ850
Uses the incoming Q.850 reason cause header and sends
the SIP reason code when the original Q.850 code is
available. When received, the reason code supersedes any
entries in the Error Mapping SIP To Cause table. The
reason code sent is not affected by the entries in the Error
Mapping SIP To Cause table.
Select how the Referred-By header is used when participating in a transfer in the Referred-By
Support drop-down menu.
Table 288: Referred-By Support Parameters
Parameter
None
Description
When acting as the transferor (sending the REFER), the
REFER does not contain a Referred-By header.
When acting as the transferee (receiving the REFER and
sending the INVITE to the target), the Referred-By header is
not copied from the REFER to the INVITE.
HeaderOnly
When acting as the transferor (sending the REFER), the
Referred-By header contains the SIP URI of the transferor.
When acting as the transferee (receiving the REFER and
sending the INVITE to the target), the Referred-By header is
copied from the REFER to the INVITE.
Dgw v2.0 Application
3.
Click Submit if you do not need to set other parameters.
4.
Set the interval, in seconds, at which SIP Keep Alive requests using SIP OPTIONS or Ping are sent
to verify the server status in the Keep Alive Interval field.
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PRACK
PRACK
Standards Supported
•
RFC 3262: Reliability of Provisional Responses in the
Session Initiation Protocol (SIP)
•
RFC 3311: The Session Initiation Protocol (SIP) UPDATE
Methoda
a. Only support receiving UPDATE. Sending an UPDATE is not supported.
The Mediatrix unit supports reliable provisional responses (PRACK) as per RFC 3262. You can define this
support when acting as a user agent client and when acting as a user agent server.
The Mediatrix unit supports the UPDATE as per RFC 3311; however, its support is limited to reception.
 To define the PRACK support:
1.
In the PRACK section of the Misc page, define the support of RFC 3262 (PRACK) when acting as
a user agent server in the UAS PRACK Support drop-down menu.
Figure 140: PRACK Section
1
2
Table 289: PRACK User Agent Server Parameters
Parameter
Description
Unsupported
The option tag “100rel” is ignored if present in the Supported
or Required header of received initial INVITEs and
provisional responses are not sent reliably as per RFC 3261.
Supported
If the option tag “100rel” is present in the Supported or
Required header of initial received INVITEs, provisional
responses are sent reliably as per RFC 3262 by adding the
option tag “100rel” to the Require header.
Receiving an UPDATE request to negotiate “early media” is supported only if you have selected
Supported.
2.
Define the support of RFC 3262 (PRACK) when acting as user agent client in the UAC PRACK
Support drop-down menu.
Table 290: PRACK User Agent Client Parameters
Parameter
310
Description
Unsupported
The option tag “100rel” is not added in the Supported or
Required header of sent INVITEs as per RFC 3261. If the
provisional response contains a Require header field with
the option tag “100rel”, the indication is ignored and no
PRACK are sent.
Supported
The option tag “100rel” is added to the Supported header of
sent initial INVITEs as per RFC 3262. If the received
provisional response contains a Require header field with
the option tag “100rel”, the response is sent reliably using
the PRACK method.
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Session Refresh
Software Configuration Guide
Table 290: PRACK User Agent Client Parameters (Continued)
Parameter
Description
Required
3.
The option tag “100rel” is added to the Require header of
sent initial INVITEs as per RFC 3262. If the received
provisional response contains a Require header field with
the option tag “100rel”, the response is sent reliably using
the PRACK method.
Click Submit if you do not need to set other parameters.
Forked Provisional Responses Behaviour
This section describes configuration that is available only in the MIB parameters of the Mediatrix unit. You can
configure these parameters as follows:



by using a MIB browser
by using the CLI
by creating a configuration script containing the configuration variables
You can configure the unit's behaviour when receiving forked provisional answers. This configuration has no
effect if the UAC PRACK Support drop-down menu is set to a value other than Unsupported.
The following values are supported:
Table 291: Forked Provisional Responses Behaviour Parameters
Value
Description
InterpretFirst
Only the first provisional answer is interpreted. Following responses do not change the
state of the call and the SDP is ignored if present.
InterpretAll
Each forked provisional response received by the unit is interpreted replacing the previous
one. If the response contains SDP, it replaces previous answers if any.
 To set the forked provisional responses behaviour:
1.
In the sipEpMIB, define the behaviour in the interopForkedProvisionalResponsesBehavior
variable.
You can also use the following line in the CLI or a configuration script:
sipEp.interopForkedProvisionalResponsesBehavior=[value]
where Value may be as follows:.
Table 292: Forked Provisional Responses Behaviour Values
Value
Meaning
100
InterpretFirst
200
InterpretAll
Session Refresh
This section allows you to define session refresh and session timers parameters. Session timers apply to the
whole unit.
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Session Refresh
 To set Session Refresh information:
1.
In the Session Refresh section of the Misc page, define whether to enable or disable the session
expiration services in the Session Refresh Timer Enable drop-down menu.
Figure 141: Session Refresh Section
1
2
1
2
Disabling this service is not recommended since it will make 'dead' calls impossible to detect.
See “Background Information” on page 312 for more details.
2.
Set the session timer minimum expiration delay, in seconds, in the Minimum Expiration Delay (s)
field.
This is the minimum value, in seconds, for the periodical session refreshes. It must be equal to or
smaller than the maximum value. This value is reflected in the Min-SE header.
The Min-SE value is a threshold under which proxies and user agents on the signalling path are not
allowed to go. Increasing the minimum helps to reduce network traffic, but also makes “dead” calls
longer to detect.
3.
Set the session timer maximum expiration delay, in seconds, in the Maximum Expiration Delay (s)
field.
This is the suggested maximum time, in seconds, for the periodical session refreshes. It must be
equal to or greater than the minimum value. This value is reflected in the Session-Expires header.
Increasing the maximum helps to reduce network traffic, but also makes “dead” calls longer to
detect.
Note: When the Maximum Expiration Delay value is lower than the Minimum Expiration Delay value, the
minimum and maximum expiration delay values in INVITE packets are the same as the value set in the
Minimum Expiration Delay field.
4.
Select the method used for sending Session Refresh Requests in the Use UPDATE for Session
Refresh parameter.
Table 293: UPDATE for Session Refresh Parameters
Parameter
Description
ReInvite
Session Refresh Requests are sent with the INVITE method.
Update
Session Refresh Requests are sent with the UPDATE
method.
Session Refresh Requests can be received via both methods, regardless of how this parameter is
configured.
5.
Click Submit if you do not need to set other parameters.
Background Information
The following explains how the session timers are used.
What is the session timer extension?
The session timer extension allows detecting the premature end of a call caused by a network problem or a
peer’s failure by resending a refresh request at every n seconds. This refresh request is either an reINVITE or
an UPDATE, according to the configuration of the Session Refresh Request Method parameter (see “PRACK”
on page 310).
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A successful response (200 OK) to this refresh request indicates that the peer is still alive and reachable. A
timeout to this refresh request may mean that there are problems in the signalling path or that the peer is no
longer available. In that case, the call is shut down by using normal SIP means.
SDP in Session Timer reINVITEs or UPDATEs
The reINVITE is sent with the last SDP that was negotiated. Receiving a session timer reINVITE should not
modify the connection characteristics.
If the reINVITE method is used, it is sent with the last SDP that was negotiated. Reception of a session timer
reINVITE should not modify the connection characteristics. If the UPDATE method is used, it is sent without
any SDP offer. REMPLACER
Relation Between Minimum and Maximum Values
A user agent that receives a Session-Expires header whose value is smaller than the minimum it is willing to
accept replies a “422 Timer too low” to the INVITE and terminates the call. The phone does not ring.
It is up to the caller to decide what to do when it receives a 422 to its INVITE. The Mediatrix unit will
automatically retry the INVITE, with a Session-Expires value equal to the minimum value that the user agent
server was ready to accept (located in the Min-SE header). This means that the maximum value as set in the
Mediatrix unit might not be followed. This has the advantageous effect of establishing the call even if the two
endpoints have conflicting values. The Mediatrix unit will also keep retrying as long as it gets 422 answers with
different Min-SE values.
Who Refreshes the Session?
Sending a session timer reINVITE or UPDATE is referred to as refreshing the session. Normally, the user
agent server that receives the INVITE has the last word on who refreshes. The Mediatrix unit always lets the
user agent client (caller) perform the refreshes if the caller supports session timers. In the case where the
caller does not support session timers, the Mediatrix unit assumes the role of the refresher.
SIP Gateway Configuration
You can define whether or not to override the SIP domain used.
 To set the SIP domain override:
1.
In the SIP Gateway Configuration section of the Misc page, define whether or not to override the
SIP domain used in the SIP Domain field.
If not empty, it overrides the home domain proxy (Proxy Host field of the Servers sub-page – SIP
Default Servers section “SIP Servers Configuration” on page 248) in the address of record and the
request-URI. When it overrides the home domain proxy in the request-URI, the request-URI also
contains a maddr parameter with the resolved home domain proxy to make sure the requests are
routable.
Figure 142: SIP Gateway Configuration Section
1
2
2.
Dgw v2.0 Application
Click Submit if you do not need to set other parameters.
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Chapter 32 - Miscellaneous SIP Parameters
SIP Blind Transfer Method
SIP Blind Transfer Method
You can set the SIP transfer method when an endpoint is acting as the transferor in a blind transfer scenario.
 To set the SIP blind transfer method:
1.
In the SIP Transfer section of the Misc page, set the Blind Transfer Method.
Figure 143: SIP Transfer Section
1
Table 294: SIP Blind Transfer Method Parameters
Parameter
2.
Description
Semi Attended
When blind transfer is invoked by the transferor, the device
sends immediately a REFER (it does not wait for the
reception of the 200OK response). This allows the call
transfer to be executed before the transfer-target answers.
The transferee and the target are then connected together
early and the transferee can hear the ringback from the
target until the target answers.
Semi Attended Confirmed
When blind transfer is invoked by the transferor, the device
waits for reception of the 200 OK from the transfer-target
before sending a REFER to the transferee.
Semi Attended Cancelled
This method is similar to the Semi Attended Transfer
method except that the INVITE sent to the transfer-target is
cancelled when the blind transfer is invoked before
receiving a 200OK (INVITE). In case where the transferor
receives a 200OK (INVITE) from the transfer-target before
receiving of a 487 Request Terminated, the transfer stays
ongoing and it behaves as a Semi Attended Confirmed
Transfer.
Click Submit if you do not need to set other parameters.
Diversion Configuration
You can define call diversion parameters.
Note: The Diversion feature is not available in the NI2 and QSIG signalling protocols. See “PRI
Configuration” on page 155 for more details on how to configure the signalling protocol.
 To set the call diversion parameters:
1.
In the Diversion section of the Misc page, set the Methcd drop-down menu with the SIP method
used to receive/send call diversion information in an INVITE.
The gateways available are those defined in “SIP Gateways Configuration” on page 243.
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Figure 144: Diversion Configuration Section
1
Table 295: Diversion Parameters
Parameter
2.
Description
None
No diversion information is sent in SIP messages.
Diversion Header
The SIP gateway supports the SIP header 'Diversion' (RFC
5806) in received and sent INVITEs, as well as in 302
messages.
Click Submit if you do not need to set other parameters.
DNS Configuration
You can define DNS-related parameters.
 To set the DNS-related parameters:
1.
In the DNS section of the Misc page, set the Supported DNS Queries drop-down menu with the type
of DNS queries that the SipEp service supports and uses.
Figure 145: DNS Configuration Section
1
Table 296: DNS Parameters
Parameter
2.
Description
Address
Sends only Address requests (type A).
SRV
Sends a Service request (type SRV) first and then Address
requests (type A) if needed.
NAPTR
Sends a Naming Authority Pointer request (type NAPTR)
first and then Service requests (type SRV) or Address
requests (type A) as needed.
Click Submit if you do not need to set other parameters.
Event Handling Configuration
The Mediatrix unit supports receiving event handling Notifications to start a remote reboot or a sync of
configuration for specific endpoint(s). The event handling Notifications "reboot" or "check-sync" is not
specified in an Allow-Events header. The Mediatrix unit supports the Notify without subscription.
It is recommended to use these event handling notifications only when the SIP transport is secure (TLS) or
when the firewall filters the requests sent to the unit.
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Event Handling Configuration
 To set the event handling parameters:
1.
In the Event Handling section of the Misc page, set the Reboot column of each available gateway
to define whether or not the SIP gateway can start a remote reboot via a SIP NOTIFY Event.
This specifies whether a remote reboot via a SIP NOTIFY message event is supported or not for a
specific SIP gateway.
Figure 146: Event Handling Parameters
1
2
Table 297: Reboot Event Handling Parameters
Parameter
2.
Description
Rejected
The "reboot" notification is rejected on reception.
Restart
When receiving a "reboot" notification, a restart of the unit is
done.
Set the CheckSync column of each available gateway to define whether or not the SIP gateway can
transfer and run a configuration file via a SIP NOTIFY Event.
This specifies whether a transfer script via a SIP NOTIFY message event is supported or not for a
specific SIP gateway.
Table 298: CheckSync Event Handling Parameters
Parameter
3.
316
Description
Rejected
The "check-sync" notification is rejected on reception.
TransferScript
When receiving a "check-sync" notification, the
Conf.ConfiguredScriptsTransferAndRun command is
executed.
Click Submit if you do not need to set other parameters.
Dgw v2.0 Application
Messaging Subscription
Software Configuration Guide
Messaging Subscription
The Mediatrix unit allows you to add the username in the Request-URI of SUBSCRIBEs it sends.
 To set the messaging subscription:
1.
In the Messaging Subscription section of the Misc page, set the Username in Request-URI dropdown menu, set whether or not the unit adds the username in the request URI of MWI SUBSCRIBE
requests.
Figure 147: Messaging Subscription Parameters
1
Table 299: Messaging Subscription Parameters
Parameter
2.
Description
Enable
The unit adds the username in the Request-URI of sent MWI
SUBSCRIBE requests.
Disable
No username in Request-URI of MWI SUBSCRIBE
requests sent by the unit.
Click Submit if you do not need to set other parameters.
Advice of Charge Configuration
The Mediatrix unit allows you to configure the Advice Of Charge (AOC) to send the current charge (D)uring a
call via an AOC-D message or the total charge at the (E)nd of call via an AOC-E message.
Note: The AOC feature is not available LP/4100/C7 Series models.
 To set the AOC parameters:
1.
In the AOC section of the Misc page, set the AOC-D Support and AOC-E Support parameters for
each available gateway to define if and how AOC-D and AOC-E messages are sent.
Figure 148: AOC-D and AOC-E Configuration Parameters
1
Table 300: AOC-D and AOC-E Configuration Parameters
Parameter
2.
Dgw v2.0 Application
Description
Disabled
No AOC information is sent. Received AOC information is
discarded.
Transparent
AOC information is forwarded to the peer interface if AOC
messages are received from the network.
Click Submit if you do not need to set other parameters.
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Chapter 32 - Miscellaneous SIP Parameters
Additional DNS Parameters
Additional DNS Parameters
This section describes configuration that is available only in the MIB parameters of the Mediatrix unit. You can
configure these parameters as follows:



by using a MIB browser
by using the CLI
by creating a configuration script containing the configuration variables
DNS Failure Concealment
You can configure the way failed DNS queries are handled.
Table 301: DNS Failure Concealment Parameters
Parameter
Description
None
When a DNS query times out or returns an error, the SIP transaction fails.
OnNoResol
ution
When a DNS query times out or returns an error, the result from the last successful query for
the same FQDN is used.
 To set the DNS failure concealment parameter:
1.
In the sipMIB, locate the DnsGroup folder.
2.
Set the DNS failure concealment configuration in the DnsFailureConcealment variable.
You can also use the following line in the CLI or a configuration script:
sip.DnsFailureConcealment="Value"
where Value may be as follows:
Table 302: DNS Failure Concealment Values
Value
Meaning
100
None
300
OnNoResolution
Note: This variable applies only to gateway type ‘Endpoint’; it has no effect on Trunk gateways, and
therefore, DNS failure concealment is always considered to be “none”.
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Media Parameters
Page Left Intentionally Blank
Codec Descriptions
C
Software Configuration Guide
H A P T E R
33
Voice & Fax Codecs
Configuration
This chapter describes the voice and fax codec configuration parameters.



Codec descriptions.
How to enable and disable the codecs.
How to set the individual codecs’ parameters.
Codec Descriptions
The Mediatrix unit supports several voice and fax codecs. It also supports unicast applications, but not
multicast ones. All voice transport is done over UDP.
All the endpoints of the Mediatrix unit can simultaneously use the same codec (for instance, G.711 PCMA), or
a mix of any of the supported codecs. Set and enable these codecs for each endpoint.
Table 303: Codecs Comparison
Compression
Voice Quality
G.711
None
Excellent
G.723.1a
Highest
Good
G.726
Medium
Fair
G.729a/ab
High
Fair/Good
a. This codec is not available on the Mediatrix C7 Series and
4102S models.
G.711 A-Law and µ-Law
The audio data is encoded as 8 bits per sample, after logarithmic scaling.
Table 304: G.711 Features
Feature
Packetization time
Description
Range of 10 ms to 30 ms with increments of 10 ms. See “G.711 Codec
Parameters” on page 328 for more details.
For the reception, the range is extended from 10 ms to 100 ms with
increments of 1 ms only if the stream is not encrypted (SRTP).
Dgw v2.0 Application
Voice Activity Detection (VAD)
Two levels of detection are available: transparent or conservative. See
“Generic Voice Activity Detection (VAD)” on page 328 for more details.
Comfort noise
Uses custom comfort noise as defined in RFC 3389.
Available for voice
Yes
Available for fax
Yes
Available for modem
Yes
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Chapter 33 - Voice & Fax Codecs Configuration
Codec Descriptions
G.723.1
Dual-rate speech coder for multimedia communications transmitting at 5.3 kbit/s and 6.3 kbit/s. This
Recommendation specifies a coded representation that can be used to compress the speech signal
component of multi-media services at a very low bit rate. The audio is encoded in 30 ms frames.
A G.723.1 frame can be one of three sizes: 24 octets (6.3 kb/s frame), 20 octets (5.3 kb/s frame), or 4 octets.
These 4-octet frames are called SID frames (Silence Insertion Descriptor) and are used to specify comfort
noise parameters.
Table 305: G.723.1 Features
Feature
Packetization time
Description
Range of 30 ms to 60 ms with increments of 30 ms. See “G.723 Codec
Parameters” on page 330 for more details.
For the reception, the range is extended from 30 ms to 120 ms with
increments of 30 ms only if the stream is not encrypted (SRTP).
Voice Activity Detection (VAD)
Supports the annex A, which is the built-in support of VAD in G.723.1.
Payload type
4
Available for voice
Yes
Available for fax
No
Available for modem
No
G.726
Algorithm recommended for conversion of a single 64 kbit/s A-law or U-law PCM channel encoded at 8000
samples/s to and from a 40, 32, 24, or 16 kbit/s channel. The conversion is applied to the PCM stream using
an Adaptive Differential Pulse Code Modulation (ADPCM) transcoding technique.
Table 306: G.726 Features
Feature
Packetization time
Description
Range of 10 ms to 30 ms with increments of 10 ms. See “G.726 Codecs
Parameters” on page 331 for more details.
For the reception, the range is extended from 10 ms to 100 ms with
increments of 1 ms only if the stream is not encrypted (SRTP).
Voice Activity Detection (VAD)
Two levels of detection are available: transparent or conservative. See
“Generic Voice Activity Detection (VAD)” on page 328 for more details.
Comfort noise
Uses custom comfort noise as defined in RFC 3389.
Payload type
Configurable as per “G.726 Codecs Parameters” on page 331.
Available for voice
Yes
Available for fax
Yes (32 kbps and 40 kbps)
Available for modem
Yes (32 kbps and 40 kbps)
G.729
Coding of speech at 8 kbit/s using conjugate structure-algebraic code excited linear prediction (CS-ACELP).
For all data rates, the sampling frequency (and RTP timestamp clock rate) is 8000 Hz.
A voice activity detector (VAD) and comfort noise generator (CNG) algorithm in Annex B of G.729 is
recommended for digital simultaneous voice and data applications; they can be used in conjunction with G.729
or G.729 Annex A. A G.729 or G.729 Annex A frame contains 10 octets, while the G.729 Annex B comfort
noise frame occupies 2 octets.
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Codec Descriptions
Software Configuration Guide
The Mediatrix unit supports G.729A and G.729AB for encoding and G.729, G.729A and G.729AB for
decoding.
Table 307: G.729 Features
Feature
Packetization time
Description
Range of 20 ms to 80 ms with increments of 10 ms. See “G.729 Codec
Parameters” on page 333 for more details.
For reception, the range is extended from 10 ms to 100 ms with
increments of 10 ms only if the stream is not encrypted (SRTP).
Voice Activity Detection (VAD)
Supports the annex B, which is the built-in support of VAD in G.729. See
“G.729 Codec Parameters” on page 333 for more details.
Payload type
18
Available for voice
Yes
Available for fax
No
Available for modem
No
Clear Mode
The Clear Mode codec is similar to the G.711 codec but without any modification of the 64 kbit/s payload (no
encoding or decoding). The Clear Mode codec thus does not have echo cancellation and a fix jitter buffer.
Clear Mode is a method to carry 64 kbit/s channel data transparently in RTP packets. This codec always uses
the RTP transport.
Table 308: Clear Mode Features
Feature
Packetization time
Description
Range of 10 ms to 30 ms with increments of 10 ms. See “Clear Mode
Codec Parameters” on page 334 for more details.
For the reception, the range is extended from 10 ms to 100 ms with
increments of 1 ms only if the stream is not encrypted (SRTP).
Voice Activity Detection (VAD)
N/A
Comfort noise
N/A
Payload type
Configurable as per “Clear Mode Codec Parameters” on page 334.
Available for voice
Yes
Available for fax
Yes
Available for modem
Yes
Clear Channel
The Clear Channel codec is similar to the G.711 codec but without any modification of the 64 kbit/s payload
(no encoding or decoding). The Clear Channel codec thus does not have echo cancellation and a fix jitter
buffer. Clear Channel is a method to carry 64 kbit/s channel data transparently in RTP packets.
The Clear Channel codec behaves like the Clear Mode codec (as defined in RFC 4040) but it uses “X-CLEARCHANNEL” mime type instead of the “CLEAR MODE” mime type during codec negotiation.
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Chapter 33 - Voice & Fax Codecs Configuration
Codec Descriptions
This codec always uses the RTP transport.
Table 309: Clear Channel Features
Feature
Packetization time
Description
Range of 10 ms to 30 ms with increments of 10 ms. See “Clear Channel
Codec Parameters” on page 335 for more details.
For the reception, the range is extended from 10 ms to 100 ms with
increments of 1 ms only if the stream is not encrypted (SRTP).
Voice Activity Detection (VAD)
N/A
Comfort noise
N/A
Payload type
Configurable as per “Clear Channel Codec Parameters” on page 335.
Available for voice
Yes
Available for fax
Yes
Available for modem
Yes
X-CCD Clear Channel
The Clear Channel codec is similar to the G.711 codec but without any modification of the 64 kbit/s payload
(no encoding or decoding). The X-CCD Clear Channel codec thus does not have echo cancellation and a fix
jitter buffer. The X-CCD Clear Channel is a method to carry 64 kbit/s channel data transparently in RTP
packets.
The X-CCD behaves like the Clear Mode codec (as defined in RFC 4040) but it uses the “X-CCD” mime type
instead of the “CLEARMODE” mime type during codec negotiation.
This codec always uses the RTP transport.
Table 310: X-CCD Clear Channel Features
Feature
Description
Packetization time
Range of 10 ms to 100 ms with increments of 1 ms. See “X-CCD Clear
Channel Codec Parameters” on page 337 for more details.
Voice Activity Detection (VAD)
N/A
Comfort noise
N/A
Payload type
Configurable as per “X-CCD Clear Channel Codec Parameters” on
page 337.
Available for voice
Yes
Available for fax
Yes
Available for modem
Yes
T.38
T.38 fax relay is a real-time fax transmission; that is, two fax machines communicating with each other as if
there were a direct phone line between the two. T.38 is called a fax relay, which means that instead of sending
inband fax signals, which implies a loss of signal quality, it sends those fax signals out-of-band in a T.38
payload, so that the remote end can reproduce the signal locally.
Table 311: T.38 Features
Feature
Packetization time
324
Description
N/A
Dgw v2.0 Application
Codec Parameters
Software Configuration Guide
Table 311: T.38 Features (Continued)
Feature
Description
Voice Activity Detection (VAD)
N/A
Payload type
N/A
Available for voice
No
Available for fax
Yes
Available for modem
No
T.38 is an unsecure protocol, thus will not be used along with secure RTP (SRTP), unless the Allow Unsecure
T.38 with Secure RTP parameter has been set to Enable. See “Chapter 34 - Security” on page 345 for more
details.
Codec Parameters
The Codec section allows you to enable or disable the codecs of the Mediatrix unit, as well as access the
codec-specific parameters.
You can use two types of configuration:


Default configurations that apply to all the endpoints of the Mediatrix unit.
Specific configurations that override the default configurations. You can define specific
configurations for each endpoint in your Mediatrix unit.
 To enable or disable the codecs:
1.
In the web interface, click the Telephony link, then the CODECS sub-link.
Figure 149: Telephony – Codecs Web Page
2
3
2.
4
5
6
Select to which endpoint (interface) you want to apply the changes in the Select Endpoint dropdown menu at the top of the window.
You have the choice between Default and the interfaces of your Mediatrix unit. The number of
interfaces available vary depending on the Mediatrix unit model you have.
You can also perform this operation in the codec-specific pages.
3.
Select whether or not you want to override one or more of the available default codecs parameters
in the Endpoint Specific column of the corresponding codec(s).
This column is available only in the specific endpoints configuration.
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Chapter 33 - Voice & Fax Codecs Configuration
Codec vs. Bearer Capabilities Mapping
You can also perform this operation in the codec-specific pages.
4.
Enable one or more codecs for voice transmission by selecting Enable in the Voice column of the
corresponding codec(s).
This indicates if the codec can be selected for voice transmission. If enabled, this codec is listed as
supported for this specific endpoint. Otherwise, it is ignored.
You can also perform this operation in the codec-specific pages.
5.
Enable one or more codecs for data transmission by selecting Enable in the Data column of the
corresponding codec(s).
This indicates if the codec can be selected for data transmission. If enabled, this codec is listed as
supported for this specific endpoint. Otherwise, it is ignored.
You can also perform this operation in the codec-specific pages.
6.
Click the
button to access the corresponding codec-specific parameters.
These parameters are described in the following sections.
7.
Click Submit if you do not need to set other parameters.
Codec vs. Bearer Capabilities Mapping
The Codec vs. Bearer Capabilities Mapping section allows you to select the codec to prioritize or select in the
outgoing INVITE when the incoming SETUP’s ITC (Information Transfer Capability) matches the configured
one. On the other hand, you can also select the ITC value to set in the outgoing SETUP’s bearer capabilities
when the incoming INVITE's codec matches the configured one.
Depending on the mapping type, a codec is prioritized or selected. Prioritized means it is the first in the list of
offered codecs when initiating a call on the IP side and Selected means it is the only codec offered.
This parameter is available only on ISDN interfaces.
You can define up to three mappings.
Note: You can also modify the selection of the preferred codec in the incoming SDP by using the the interop
parameter Codec vs Beareer Capabilities Mapping Preferred Codec Choice in the SIP > Interop > SDP
section. See “SDP Interop” on page 282 for more details.
 To set the codec vs. bearer capabilities mapping
1.
Define if the outgoing codec’s priority should reflect the incoming ITC and vice versa in the Enable
column drop-down menu.
Figure 150: Codec vs. Bearer Capabilities Mapping Section
1
2
3
4
Table 312: Outgoing Codec Priority
Parameter
Disable
326
Description
The mapping is not applied:
•
The codec’s order in the outgoing INVITE follows the codec priority.
•
The ITC in the outgoing SETUP is the default one (3.1 kHz) unless a
mapping in the Call Routing table modifies it.
Dgw v2.0 Application
Codec vs. Bearer Capabilities Mapping
Software Configuration Guide
Table 312: Outgoing Codec Priority (Continued)
Parameter
Enable
Description
If the ITC value in the incoming SETUP matches the value of the corresponding
ITC drop-down menu, the first codec in the outgoing INVITE is the one set in the
CODEC drop-down menu.
If the first codec in the incoming INVITE matches the value set in the CODEC
drop-down menu, the ITC value in the outgoing SETUP is the one set in the ITC
drop-down menu.
2.
Select a codec in the CODEC column drop-down menu.
This is the codec to be prioritized or selected in an outgoing INVITE when the incoming SETUP's
ITC matches the value set in the corresponding ITC drop-down menu. This codec is also checked
against an incoming INVITE's priority codec. If it matches, then the outgoing SETUP's ITC is set to
the value in the corresponding ITC drop-down menu.
See Step 3 for a description of prioritization versus selection of a codec.
3.
Set the mapping type of the codec in the Mapping Type drop-down value.
Table 313: Codec Mapping Type
Parameter
4.
Description
Prioritize
The codec is set on top of the list in an outgoing INVITE when the incoming
SETUP's ITC matches the value set in the corresponding ITC drop-down menu.
Select
The codec is the only one offered in an outgoing INVITE when the incoming
SETUP's ITC matches the value set in the corresponding ITC drop-down menu.
Select an ITC value in the ITC column drop-down menu.
This is the ITC value to be set in the outgoing SETUP when the incoming INVITE's priority codec
matches the value of the corresponding CODEC drop-down menu. This value is also checked
against an incoming SETUP's bearer capabilities. If it matches, then the outgoing INVITE's
prioritized or selected codec is set to the value of the corresponding CODEC drop-down menu.
Table 314: Information Transfer Capability Values
Value
Description
speech
Voice terminals (telephones).
unrestricted
Unrestricted digital information (64 kbps).
3.1Khz
Transparent 3.1 kHz audio channel.
For an incoming ISDN call using UDI (Unrestricted Digital), the Codec vs Bearer Capabilities
Mapping entries with the ITC set to Unrestricted is used only if the codec is Clear Mode, Clear
Channel, XCCD, G.711 a-law or G.711 u-law. G.711 a-law and G.711 u-law are also used only if
they match the ISDN port Preferred Encoding Scheme value (see “PRI Configuration” on page 155
and “BRI Configuration” on page 167 for more details).
See Step 3 for a description of prioritization versus selection of a codec.
5.
Dgw v2.0 Application
Click Submit if you do not need to set other parameters.
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Chapter 33 - Voice & Fax Codecs Configuration
Generic Voice Activity Detection (VAD)
Generic Voice Activity Detection (VAD)
VAD defines how the Mediatrix unit sends information pertaining to silence. This allows the unit to detect when
the user talks, thus avoiding to send silent RTP packets. This saves on network resources. However, VAD
may affect packets that are not really silent (for instance, cut sounds that are too low). VAD can thus slightly
affect the voice quality.
 To set the generic Voice Activity Detection (VAD)
1.
In the Generic Voice Activity Detection (VAD) section, select whether or not you want to override
the VAD parameters set in the Default configuration in the Endpoint Specific drop-down menu.
This menu is available only in the specific endpoints configuration.
Figure 151: Generic Voice Activity Detection (VAD) Section
1
2.
2
Enable the G.711 and G.726 Voice Activity Detection (VAD) by selecting the proper setting in the
Enable (G711 and G726) drop-down menu.
Table 315: G.711/G.726 VAD Settings
Setting
Description
Disable
VAD is not used.
Transparent
VAD is enabled. It has low sensitivity to silence periods.
Conservative
VAD is enabled. It has normal sensitivity to silence periods.
The difference between transparent and conservative is how “aggressive” the algorithm considers
something as an inactive voice and how “fast” it stops the voice stream. A setting of conservative is
a little bit more aggressive to react to silence compared to a setting of transparent.
3.
Click Submit if you do not need to set other parameters.
G.711 Codec Parameters
The following are the G.711 codec parameters you can set. There are two sections for G.711:


G.711 a-law
G.711 u-law
These sections use the same parameters, so only one of them is described below.
 To set the G.711 codec parameters:
1.
In the CODEC section of the CODECS page, click the
G.711 codec to access the codec-specific parameters.
button at the right of the corresponding
2.
Select to which endpoint (interface) you want to apply the changes in the Select Endpoint dropdown menu at the top of the window.
You have the choice between Default and the interfaces of your Mediatrix unit. The number of
interfaces available vary depending on the Mediatrix unit model you have.
328
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G.711 Codec Parameters
Software Configuration Guide
Figure 152: G.711 a-law Section
2
3
4
5
6
7
8
3.
Select whether or not you want to override the G.711 parameters set in the Default configuration in
the Use Endpoint Specific drop-down menu.
This menu is available only in the specific endpoints configuration.
You can also perform this operation in the main CODEC section.
4.
Enable the G.711 codec for voice transmission by selecting Enable in the Voice Transmission dropdown menu.
This indicates if the codec can be selected for voice transmission. If enabled, this codec is listed as
supported for this specific endpoint. Otherwise, it is ignored.
You can also perform this operation in the main CODEC section.
5.
Set the default priority for voice in the Voice Priority field.
This sets the priority between different codecs. Codecs with a higher priority are used first, a priority
of 0 being the lowest priority. For instance, a codec with priority 3 is used before a codec with priority
2. The maximum priority is 10.
The Mediatrix unit uses an internal order for codecs with the same priority.
Note: The codec used is also related to the SIP negotiation. The priority order affects the SIP negotiation,
which decides on the codec to use.
6.
Enable the G.711 codec for data transmission by selecting Enable in the Data Transmission dropdown menu.
This indicates if the codec can be selected for data transmission. If enabled, this codec is listed as
supported for this specific endpoint. Otherwise, it is ignored.
You can also perform this operation in the main CODEC section.
7.
Set the default priority for data in the Data Priority field.
This sets the priority between different codecs. Codecs with a higher priority are used first, a priority
of 0 being the lowest priority. For instance, a codec with priority 3 is used before a codec with priority
2. The maximum priority is 10.
The Mediatrix unit uses an internal order for codecs with the same priority.
8.
Select the minimum and maximum packetization time values for the codec in the Minimum
Packetization Time and Maximum Packetization Time drop-down menus.
The packetization time (also called packetization period or ptime) is the duration, in ms, of the voice
packet. The range is from 10 ms to 30 ms with increments of 10 ms.
For the reception, the range is extended from 10 ms to 100 ms with increments of 1 ms only if the
stream is not encrypted (SRTP).
9.
Dgw v2.0 Application
Click Submit if you do not need to set other parameters.
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Chapter 33 - Voice & Fax Codecs Configuration
G.723 Codec Parameters
You can also access the specific parameters of another codec by selecting the codec in the Select
CODEC drop-down menu at the top of the page.
G.723 Codec Parameters
The following are the G.723 codec parameters you can set.
Note that the G.723 codec is not available on the Mediatrix C7 Series and 4102S models.
 To set the G.723 codec parameters:
1.
In the CODEC section of the CODECS page, click the
to access the codec-specific parameters.
button at the right of the G.723 codec
2.
Select to which endpoint (interface) you want to apply the changes in the Select Endpoint dropdown menu at the top of the window.
You have the choice between Default and the interfaces of your Mediatrix unit. The number of
interfaces available vary depending on the Mediatrix unit model you have.
Figure 153: G.723 Section
2
3
4
5
6
7
3.
Select whether or not you want to override the G.723 parameters set in the Default configuration in
the Use Endpoint Specific drop-down menu.
This menu is available only in the specific endpoints configuration.
You can also perform this operation in the main CODEC section.
4.
Enable the G.723 codec for voice transmission by selecting Enable in the Voice Transmission dropdown menu.
This indicates if the codec can be selected for voice transmission. If enabled, this codec is listed as
supported for this specific endpoint. Otherwise, it is ignored.
You can also perform this operation in the main CODEC section.
5.
Set the default priority for voice in the Voice Priority field.
This sets the priority between different codecs. Codecs with a higher priority are used first, a priority
of 0 being the lowest priority. For instance, a codec with priority 3 is used before a codec with priority
2. The maximum priority is 10.
The Mediatrix unit uses an internal order for codecs with the same priority.
Note: The codec used is also related to the SIP negotiation. The priority order affects the SIP negotiation,
which decides on the codec to use.
6.
Select the G.723 bit rate in the Bit Rate drop-down menu.
You have the following choices:
•
330
53 Kbs
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G.726 Codecs Parameters
Software Configuration Guide
•
7.
63 Kbs
Select the minimum and maximum packetization time values for the codec in the Minimum
Packetization Time and Maximum Packetization Time drop-down menus.
The packetization time (also called packetization period or ptime) is the duration, in ms, of the voice
packet. The range is from 30 ms to 60 ms with increments of 30 ms.
For the reception, the range is extended from 30 ms to 120 ms with increments of 30 ms only if the
kstream is not encrypted (SRTP).
8.
Click Submit if you do not need to set other parameters.
You can also access the specific parameters of another codec by selecting the codec in the Select
CODEC drop-down menu at the top of the page.
G.726 Codecs Parameters
The following are the G.726 codecs parameters you can set. There are four sections for G.726:




G.726 16 Kbps
G.726 24 Kbps
G.726 32 Kbps
G.726 40 Kbps
These sections offer almost the same parameters, except that you cannot use the G.726 16 Kbps and
G.726 24 Kbps codecs for fax transmission.
 To set the G.726 codecs parameters:
1.
In the CODEC section of the CODECS page, click the
G.726 codec to access the codec-specific parameters.
button at the right of the corresponding
2.
Select to which endpoint (interface) you want to apply the changes in the Select Endpoint dropdown menu at the top of the window.
You have the choice between Default and the interfaces of your Mediatrix unit. The number of
interfaces available vary depending on the Mediatrix unit model you have.
Figure 154: G.726 Section
2
3
4
5
6
7
8
9
3.
Select whether or not you want to override the G.726 parameters set in the Default configuration in
the Use Endpoint Specific drop-down menu.
This menu is available only in the specific endpoints configuration.
You can also perform this operation in the main CODEC section.
4.
Dgw v2.0 Application
Enable the corresponding G.726 codec for voice transmission by selecting Enable in the Voice
Transmission drop-down menu.
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G.726 Codecs Parameters
This indicates if the codec can be selected for voice transmission. If enabled, this codec is listed as
supported for this specific endpoint. Otherwise, it is ignored.
You can also perform this operation in the main CODEC section.
5.
Set the default priority for voice in the Voice Priority field.
This sets the priority between different codecs. Codecs with a higher priority are used first, a priority
of 0 being the lowest priority. For instance, a codec with priority 3 is used before a codec with priority
2. The maximum priority is 10.
The Mediatrix unit uses an internal order for codecs with the same priority.
Note: The codec used is also related to the SIP negotiation. The priority order affects the SIP negotiation,
which decides on the codec to use.
6.
Enable the codec for data transmission by selecting Enable in the Data Transmission drop-down
menu.
This indicates if the codec can be selected for data transmission. If enabled, this codec is listed as
supported for this specific endpoint. Otherwise, it is ignored.
This menu is not available for the G.726 16 Kbps and G.726 24 Kbps codecs.
You can also perform this operation in the main CODEC section.
7.
Set the default priority for data in the Data Priority field.
This sets the priority between different codecs. Codecs with a higher priority are used first, a priority
of 0 being the lowest priority. For instance, a codec with priority 3 is used before a codec with priority
2. The maximum priority is 10.
The Mediatrix unit uses an internal order for codecs with the same priority.
This field is not available for the G.726 16 Kbps and G.726 24 Kbps codecs.
8.
Set the G.726 actual RTP dynamic payload type used in an initial offer in the Payload Type field.
The payload types available are as per RFC 3551. The values range from 96 to 127. The default
values are as follows:
Table 316: G.726 Default Payload Type
Codec
9.
Default Value
G.726 (16 kbps)
97
G.726 (24 kbps)
98
G.726 (32 kbps)
99
G.726 (40 kbps)
100
Select the minimum and maximum packetization time values for the G.726 codec in the Minimum
Packetization Time and Maximum Packetization Time drop-down menus.
The packetization time (also called packetization period or ptime) is the duration, in ms, of the voice
packet. The range is from 10 ms to 30 ms with increments of 10 ms.
For the reception, the range is extended from 10 ms to 100 ms with increments of 1 ms only if the
stream is not encrypted (SRTP).
10.
Click Submit if you do not need to set other parameters.
You can also access the specific parameters of another codec by selecting the codec in the Select
CODEC drop-down menu at the top of the page.
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G.729 Codec Parameters
Software Configuration Guide
G.729 Codec Parameters
The following are the G.729 codec parameters you can set.
 To set the G.729 codec parameters:
1.
In the CODEC section of the CODECS page, click the
to access the codec-specific parameters.
button at the right of the G.729 codec
2.
Select to which endpoint (interface) you want to apply the changes in the Select Endpoint dropdown menu at the top of the window.
You have the choice between Default and the interfaces of your Mediatrix unit. The number of
interfaces available vary depending on the Mediatrix unit model you have.
Figure 155: G.729 Section
2
3
4
5
6
7
3.
In the G.729 section, select whether or not you want to override the G.729 parameters set in the
Default configuration in the Use Endpoint Specific drop-down menu.
This menu is available only in the specific endpoints configuration.
You can also perform this operation in the main CODEC section.
4.
Enable the G.729 codec for voice transmission by selecting Enable in the Voice Transmission dropdown menu.
This indicates if the codec can be selected for voice transmission. If enabled, this codec is listed as
supported for this specific endpoint. Otherwise, it is ignored.
You can also perform this operation in the main CODEC section.
5.
Set the default priority for voice in the Voice Priority field.
This sets the priority between different codecs. Codecs with a higher priority are used first, a priority
of 0 being the lowest priority. For instance, a codec with priority 3 is used before a codec with priority
2. The maximum priority is 10.
The Mediatrix unit uses an internal order for codecs with the same priority.
Note: The codec used is also related to the SIP negotiation. The priority order affects the SIP negotiation,
which decides on the codec to use.
6.
Select the minimum and maximum packetization time values for the codec in the Minimum
Packetization Time and Maximum Packetization Time drop-down menus.
The packetization time (also called packetization period or ptime) is the duration, in ms, of the voice
packet. The range is from 20 ms to 80 ms with increments of 10 ms.
For reception, the range is extended from 10 ms to 100 ms with increments of 10 ms only if the
stream is not encrypted (SRTP).
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7.
Clear Mode Codec Parameters
Select the G.729 Voice Activity Detection (VAD) in the Built-in Voice Activity Detection (VAD) dropdown menu.
Table 317: G.729 VAD
Parameter
Description
Disable
G.729 uses annex A only.
Enable
G.729 annex A is used with annex B. Speech frames are only sent during talkspurts
(periods of audio activity). During silence periods, no speech frames are sent, but
Comfort Noise (CN) packets containing information about background noise may
be sent in accordance with annex B of G.729.
VAD defines how the Mediatrix unit sends information pertaining to silence. This allows the unit to
detect when the user talks, thus avoiding to send silent RTP packets. This saves on network
resources. However, VAD may affect packets that are not really silent (for instance, cut sounds that
are too low). VAD can thus slightly affect the voice quality.
G.729 has a built-in VAD in its Annex B version. It is recommended for digital simultaneous voice
and data applications and can be used in conjunction with G.729 or G.729 Annex A. A G.729 or
G.729 Annex A frame contains 10 octets, while the G.729 Annex B frame occupies 2 octets. The
CN packets are sent in accordance with annex B of G.729.
8.
Click Submit if you do not need to set other parameters.
You can also access the specific parameters of another codec by selecting the codec in the Select
CODEC drop-down menu at the top of the page.
Clear Mode Codec Parameters
The following are the Clear Mode codec parameters you can set.
 To set the Clear Mode codec parameters:
1.
In the CODEC section of the CODECS page, click the
codec to access the codec-specific parameters.
button at the right of the Clear Mode
2.
Select to which endpoint (interface) you want to apply the changes in the Select Endpoint dropdown menu at the top of the window.
You have the choice between Default and the interfaces of your Mediatrix unit. The number of
interfaces available vary depending on the Mediatrix unit model you have.
Figure 156: Clear Mode Section
2
3
4
5
6
7
8
9
3.
334
Select whether or not you want to override the Clear Mode parameters set in the Default
configuration in the Use Endpoint Specific drop-down menu.
Dgw v2.0 Application
Clear Channel Codec Parameters
Software Configuration Guide
This menu is available only in the specific endpoints configuration.
You can also perform this operation in the main CODEC section.
4.
Enable the Clear Mode codec for voice transmission by selecting Enable in the Voice Transmission
drop-down menu.
This indicates if the codec can be selected for voice transmission. If enabled, this codec is listed as
supported for this specific endpoint. Otherwise, it is ignored.
You can also perform this operation in the main CODEC section.
5.
Set the default priority for voice in the Voice Priority field.
This sets the priority between different codecs. Codecs with a higher priority are used first, a priority
of 0 being the lowest priority. For instance, a codec with priority 3 is used before a codec with priority
2. The maximum priority is 10.
The Mediatrix unit uses an internal order for codecs with the same priority.
Note: The codec used is also related to the SIP negotiation. The priority order affects the SIP negotiation,
which decides on the codec to use.
6.
Enable the Clear Mode codec for data transmission by selecting Enable in the Data Transmission
drop-down menu.
This indicates if the codec can be selected for data transmission. If enabled, this codec is listed as
supported for this specific endpoint. Otherwise, it is ignored.
You can also perform this operation in the main CODEC section.
7.
Set the default priority for data in the Data Priority field.
This sets the priority between different codecs. Codecs with a higher priority are used first, a priority
of 0 being the lowest priority. For instance, a codec with priority 3 is used before a codec with priority
2. The maximum priority is 10.
The Mediatrix unit uses an internal order for codecs with the same priority.
8.
Set the Clear Mode RTP dynamic payload type used in an initial offer in the Payload Type field.
The payload types available are as per RFC 3551. The values range from 96 to 127. The default
value is 125.
9.
Select the minimum and maximum packetization time values for the codec in the Minimum
Packetization Time and Maximum Packetization Time drop-down menus.
The packetization time (also called packetization period or ptime) is the duration, in ms, of the voice
packet. The range is from 10 ms to 30 ms with increments of 10 ms.
For the reception, the range is extended from 10 ms to 100 ms with increments of 1 ms only if the
stream is not encrypted (SRTP).
10.
Click Submit if you do not need to set other parameters.
You can also access the specific parameters of another codec by selecting the codec in the Select
CODEC drop-down menu at the top of the page.
Clear Channel Codec Parameters
The following are the Clear Channel codec parameters you can set.
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Clear Channel Codec Parameters
 To set the Clear Channel codec parameters:
1.
In the CODEC section of the CODECS page, click the
codec to access the codec-specific parameters.
button at the right of the Clear Channel
2.
Select to which endpoint (interface) you want to apply the changes in the Select Endpoint dropdown menu at the top of the window.
You have the choice between Default and the interfaces of your Mediatrix unit. The number of
interfaces available vary depending on the Mediatrix unit model you have.
Figure 157: Clear Channel Section
2
3
4
5
6
7
8
9
3.
Select whether or not you want to override the Clear Channel parameters set in the Default
configuration in the Use Endpoint Specific drop-down menu.
This menu is available only in the specific endpoints configuration.
You can also perform this operation in the main CODEC section.
4.
Enable the Clear Channel codec for voice transmission by selecting Enable in the Voice
Transmission drop-down menu.
This indicates if the codec can be selected for voice transmission. If enabled, this codec is listed as
supported for this specific endpoint. Otherwise, it is ignored.
You can also perform this operation in the main CODEC section.
5.
Set the default priority for voice in the Voice Priority field.
This sets the priority between different codecs. Codecs with a higher priority are used first, a priority
of 0 being the lowest priority. For instance, a codec with priority 3 is used before a codec with priority
2. The maximum priority is 10.
The Mediatrix unit uses an internal order for codecs with the same priority.
Note: The codec used is also related to the SIP negotiation. The priority order affects the SIP negotiation,
which decides on the codec to use.
6.
Enable the Clear Channel codec for data transmission by selecting Enable in the Data
Transmission drop-down menu.
This indicates if the codec can be selected for data transmission. If enabled, this codec is listed as
supported for this specific endpoint. Otherwise, it is ignored.
You can also perform this operation in the main CODEC section.
7.
Set the default priority for data in the Data Priority field.
This sets the priority between different codecs. Codecs with a higher priority are used first, a priority
of 0 being the lowest priority. For instance, a codec with priority 3 is used before a codec with priority
2. The maximum priority is 10.
The Mediatrix unit uses an internal order for codecs with the same priority.
336
Dgw v2.0 Application
X-CCD Clear Channel Codec Parameters
8.
Software Configuration Guide
Set the Clear Channel RTP dynamic payload type used in an initial offer in the Payload Type field.
The payload types available are as per RFC 3551. The values range from 96 to 127. The default
value is 125.
9.
Select the minimum and maximum packetization time values for the codec in the Minimum
Packetization Time and Maximum Packetization Time drop-down menus.
The packetization time (also called packetization period or ptime) is the duration, in ms, of the voice
packet. The range is from 10 ms to 30 ms with increments of 10 ms.
For the reception, the range is extended from 10 ms to 100 ms with increments of 1 ms only if the
stream is not encrypted (SRTP).
10.
Click Submit if you do not need to set other parameters.
You can also access the specific parameters of another codec by selecting the codec in the Select
CODEC drop-down menu at the top of the page.
X-CCD Clear Channel Codec Parameters
The following are the X-CCD Clear Channel codec parameters you can set.
 To set the Clear Channel codec parameters:
1.
In the CODEC section of the CODECS page, click the
to access the codec-specific parameters.
button at the right of the X CCD codec
2.
Select to which endpoint (interface) you want to apply the changes in the Select Endpoint dropdown menu at the top of the window.
You have the choice between Default and the interfaces of your Mediatrix unit. The number of
interfaces available vary depending on the Mediatrix unit model you have.
Figure 158: X CCD Section
2
3
4
5
6
7
8
9
3.
Select whether or not you want to override the X CCD parameters set in the Default configuration
in the Endpoint Specific drop-down menu.
This menu is available only in the specific endpoints configuration.
You can also perform this operation in the main CODEC section.
4.
Enable the X CCD codec for voice transmission by selecting Enable in the Voice Transmission
drop-down menu.
This indicates if the codec can be selected for voice transmission. If enabled, this codec is listed as
supported for this specific endpoint. Otherwise, it is ignored.
You can also perform this operation in the main CODEC section.
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Chapter 33 - Voice & Fax Codecs Configuration
5.
Fax Parameters
Set the default priority for voice in the Voice Priority field.
This sets the priority between different codecs. Codecs with a higher priority are used first, a priority
of 0 being the lowest priority. For instance, a codec with priority 3 is used before a codec with priority
2. The maximum priority is 10.
The Mediatrix unit uses an internal order for codecs with the same priority.
Note: The codec used is also related to the SIP negotiation. The priority order affects the SIP negotiation,
which decides on the codec to use.
6.
Enable the X CCD codec for data transmission by selecting Enable in the Data Transmission dropdown menu.
This indicates if the codec can be selected for data transmission. If enabled, this codec is listed as
supported for this specific endpoint. Otherwise, it is ignored.
You can also perform this operation in the main CODEC section.
7.
Set the default priority for data in the Data Priority field.
This sets the priority between different codecs. Codecs with a higher priority are used first, a priority
of 0 being the lowest priority. For instance, a codec with priority 3 is used before a codec with priority
2. The maximum priority is 10.
The Mediatrix unit uses an internal order for codecs with the same priority.
Note: The codec used is also related to the SIP negotiation. The priority order affects the SIP negotiation,
which decides on the codec to use.
8.
Set the X CCD RTP dynamic payload type used in an initial offer in the Payload Type field.
The payload types available are as per RFC 3551. The values range from 96 to 127. The default
value is 125.
9.
Select the minimum and maximum packetization time values for the codec in the Minimum
Packetization Time and Maximum Packetization Time drop-down menus.
The packetization time (also called packetization period or ptime) is the duration, in ms, of the voice
packet. The range is from 10 ms to 30 ms with increments of 10 ms.
10.
Click Submit if you do not need to set other parameters.
You can also access the specific parameters of another codec by selecting the codec in the Select
CODEC drop-down menu at the top of the page.
Fax Parameters
The Mediatrix unit handles G3 fax transmissions at speeds up to 14.4 kbps. Automatic fax mode detection is
standard on all endpoints. Real-Time Fax Over UDP with the T.38 protocol stack is also available.
A fax call works much like a regular voice call, with the following differences:
1.
The fax codec may be re-negotiated by using a re-INVITE.
2.
The goal of the re-INVITE is to allow both user agents to agree on a fax codec, which is either:
3.
a.
Clear channel (G.711 or G.726) without Echo Cancellation nor Silence Suppression
(automatically disabled).
b.
T.38.
Upon fax termination, if the call is not BYE, the previous voice codec is recovered with another reINVITE.
All endpoints of the Mediatrix unit can simultaneously use the same codec (for instance, T.38), or a mix of any
of the supported codecs. Set and enable these codecs for each endpoint.
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Software Configuration Guide
Clear Channel Fax
The Mediatrix unit can send faxes in clear channel. The following is a clear channel fax call flow:
Figure 159: Clear Channel Fax Call Flow
INVITE
[…]
m=audio 5006 RTP/AVP 18 0 13
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
[…]
m=audio 5004 RTP/AVP 18 0 13
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
200 OK
Ringing/Trying
ACK
RTP=G.729 (Voice Call)
Fax Tone Detected
User
Agent
#1
RTP=PCMU (Echo Cancellation + Silence Suppression = disabled)
User
Agent
#2
No re-INVITE!!
 There is no need for a re-INVITE since the far end already supports the
data codec (PCMU).
 When your SDP capabilities are inserted in a SIP packet, it implies that
you can receive any of these capabilities at any given time without notice.
 In this case, both ends should switch to clear channel automatically upon
detection of the fax transmission.
Fax is terminated
BYE
200 OK
DSP Limitation
The Mediatrix unit currently suffers from a limitation of its DSP. Because of this limitation, the voice does not
switch back to the original negotiated codec after a clear channel fax is performed.
The Mediatrix unit cannot detect the end of a clear channel fax, which means that the unit cannot switch back
to the original negotiated codec if this codec was not a clear channel codec, e.g., a session established in
G.729.
When the unit detects a fax, it automatically switches to a negotiated clear channel codec such as PCMU (if
there is no T.38 or if T.38 negotiation failed). Once the fax is terminated, the Mediatrix unit is not notified by
the DSP. The unit thus stays in the clear channel codec and does not switch back to G.729.
T.38 Fax
The Mediatrix unit can send faxes in T.38 mode over UDP. T.38 is used for fax if both units are T.38 capable;
otherwise, transmission in clear channel over G.711 as defined is used (if G.711 µ-law and/or G.711 A-law are
enabled). If no clear channel codecs are enabled and the other endpoint is not T.38 capable, the fax
transmission fails.
Caution: The Mediatrix unit opens the T.38 channel only after receiving the “200 OK” message from the
peer. This means that the Mediatrix unit cannot receive T.38 packets before receiving the “200 OK”. Based
on RFC 3264, the T.38 channel should be opened as soon as the unit sends the “INVITE” message.
The quality of T.38 fax transmissions depends upon the system configuration, type of call control system used,
type of Mediatrix units deployed, as well as the model of fax machines used. Should some of these conditions
be unsatisfactory, performance of T.38 fax transmissions may vary and be reduced below expectations.
Note: Media5 recommends not to use a fax that does not send a CNG tone. If you use such a fax to send
a fax communication to the public network, this might result in a communication failure.
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Fax Parameters
The following is a T.38 fax call flow:
Figure 160: T.38 Fax Call Flow
INVITE
[…]
m=audio 5004 RTP/AVP 0 18 4 8 13
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
[…]
m=audio 5006 RTP/AVP 0 18 4 8 13
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
Ringing/Trying
200 OK
ACK
Fax Tone Detected
INVITE
User
Agent
#1
[…]
m=image 6006 udptl t38
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxVersion:0
a=T38FaxUdpEC:t38UDPRedundancy
200 OK
Trying
[…]
m=image 6006 udptl t38
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxVersion:0
a=T38FaxUdpEC:t38UDPRedundancy
User
Agent
#2
ACK
Fax is terminated
[…]
m=audio 5004 RTP/AVP 0 18 4 8 13
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
INVITE
200 OK
Trying
[…]
m=audio 5006 RTP/AVP 0 18 4 8 13
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
ACK
BYE
200 OK
T.38 Parameters Configuration
The following are the T.38 codec parameters you can set.
 To set the T.38 codec parameters:
1.
In the CODEC section of the CODECS page, click the
G.726 codec to access the codec-specific parameters.
button at the right of the corresponding
2.
Select to which endpoint (interface) you want to apply the changes in the Select Endpoint dropdown menu at the top of the window.
You have the choice between Default and the interfaces of your Mediatrix unit. The number of
interfaces available vary depending on the Mediatrix unit model you have.
Figure 161: T.38 Section
2
4
6
8
5
7
9
10
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Software Configuration Guide
3.
In the T.38 section, select whether or not you want to override the T.38 parameters set in the Default
configuration in the Use Endpoint Specific drop-down menu.
This menu is available only in the specific endpoints configuration.
You can also perform this operation in the main CODEC section.
4.
Enable the T.38 codec by selecting Enable in the Enable drop-down menu.
You can also perform this operation in the main CODEC section.
5.
Set the default priority for fax in the Priority field.
This sets the priority between different codecs. Codecs with a higher priority are used first, a priority
of 0 being the lowest priority. For instance, a codec with priority 3 is used before a codec with priority
2. The maximum priority is 10.
The Mediatrix unit uses an internal order for codecs with the same priority.
Note: Currently, the only T.38 priority accepted is 10. Priority between 1 and 9 is refused.
6.
Set the number of redundancy packets sent with the current packet in the Redundancy Level field.
This is the standard redundancy offered by T.38. Available values range from 1 to 5. Please see
step 7 for additional reliability options for T.38.
7.
Set the T.38 input signal detection threshold in the Detection Threshold drop-down menu.
Lowering the threshold allows detecting lower amplitude fax signals. The following values are
available:
8.
•
Default: (-26 dB)
•
Low: (-31 dB)
•
Lowest: (-43 dB)
For additional reliability, define the number of times T.38 packets are retransmitted in the Frame
Redundancy Level field.
This field is available only in the default endpoint configuration.
This only applies to the T.38 packets where the PrimaryUDPTL contains the following T.38 data
type:
9.
•
HDLC_SIG_END,
•
HDLC_FCS_OK_SIG_END,
•
HDLC_FCS_BAD_SIG_END and
•
T4_NON_ECM_SIG_END
Define whether or not the Mediatrix unit sends no-signal packets during a T.38 fax transmission in
the No Signal drop-down menu.
This menu is available only in the default endpoint configuration.
When enabled, the unit ensures that, during a T.38 fax transmission, data is sent out at least every
time the No Signal Timeout delay expires. The Mediatrix unit sends no-signal packets if no
meaningful data have been sent for a user-specified period of time.
10.
Set the period, in seconds, at which no-signal packets are sent during a T.38 transmission in the No
Signal Timeout field.
This field is available only in the default endpoint configuration.
No-signal packets are sent out if there are no valid data to send.
11.
Click Submit if you do not need to set other parameters.
You can also access the specific parameters of another codec by selecting the codec in the Select
CODEC drop-down menu at the top of the page.
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Fax Parameters
Data Codec Selection Procedure
The Mediatrix unit follows a procedure when selecting data codec. This procedure is the default behaviour of
the Mediatrix unit. Some interop variables may modify this procedure. Tones are detected on the analog ports
only.
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Software Configuration Guide
Figure 162: Data Codec Selection Procedure
Voice /
Voiceband
data call
Complete fax/modem codec
selection procedure .
Tones are detected on the telephony ports except for
the CED, which can also be detected on the IP side
CNG tone
detected?
CED tone
detected?
No
No
V.21 tone
detected?
No
Continue call
Yes
Yes
Evaluate
BehaviorOnCed
ToneDetection
Stop sending
voice codec. Start
buffering T.38
packets
Faxmode
Yes
Passthrough
T.38 is
enabled?
Yes
Send a re-INVITE
for T.38
Current voice
codec is data capable?
No
Yes
Continue call with
current codec
No
T.38 accepted
by peer?
Evaluate
BehaviorOnT38Invite
NotAccepted.Behavior
according to error
code
No
List of
negotiated
codecs
contains a
data-capable
codec?
Yes
ReInviteForClear
ChannelOnly (default)
Start sending T.38
packets
At least one data
codec is enabled
on the device ?
Continue T.38 fax
DropCall
Switch to highest
negotiated priority
data codec and
continue call
No
Switch to highest
configured priority G.711
data codec
(PCMU if all disabled )
ReInviteOn
NoNegotiated
DataCodec
No
Yes
Yes
ReEstablish
Audio
Evaluate
interopB ehavior
OnMachine
Detection
ReInviteOn
FaxT38Only
Continue call
UsePrevious
MediaNegiciation
Send SIP reINVITE for datacapable codecs
Terminate call
No
Disable fax/modem detection
and send SIP re-INVITE for
voice -capable codecs
New codec
accepted by
peer ?
Yes
Restore the previouly
used voice codec and
continue call
Dgw v2.0 Application
Switch to
selected data
codec and
continue call
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Fax Parameters
Dgw v2.0 Application
Introduction
C
Software Configuration Guide
H A P T E R
34
Security
This chapter describes how to properly configure the security parameters of the Mediatrix unit.
Introduction
You can define security features on the Mediatrix unit. This section applies to media security parameters.
Applying security on the Mediatrix unit involves several steps:

Properly set the time on the Mediatrix unit by configuring a valid SNTP server (“SNTP
Configuration” on page 57) and time zone (“Time Configuration” on page 58).

Transfer a valid CA certificate into the Mediatrix unit (“Chapter 49 - Certificates Management”
on page 501).

Use secure signalling by enabling the TLS transport protocol (“Chapter 30 - SIP Transport
Parameters” on page 271).
Caution: If you enable Secure RTP (SRTP) on at least one line, it is acceptable to have the secure SIP
transport (TLS) disabled for testing purposes. However, you must never use this configuration in a
production environment, since an attacker could easily break it. Enabling TLS for SIP Transport is strongly
recommended and is usually mandatory for security interoperability with third-party equipments.
Caution: When using a codec other than G.711, enabling Secure RTP (SRTP) has an impact on the
Mediatrix unit’s overall performance as SRTP requires CPU power. The more lines use SRTP, the more
overall performance is affected. This is especially true with the Mediatrix 4116, LP16, 4124 and LP24
models. This could mean that a user picking up a telephone on these models may not have a dial tone due
to lack of resources. See also “DSP Limitation” on page 397 for more details on resources limitations with
SRTP and conferences.

Use secure media by:
•
Defining the SRTP/ SRTCP base port (“Base Ports Configuration” on page 365).
•
Setting the RTP secure mode to “Secure” or “Secure with fallback” (this section).
Security Parameters
The Security section allows you to secure the RTP stream (media) of the Mediatrix unit.
Since the SRTP encryption and authentication needs more processing, the number of calls that the Mediatrix
unit can handle simultaneously may be reduced, depending of the codecs enabled. You could set the
Mediatrix unit not to impact the number of simultaneous calls by enabling only G.711 codecs and disabling
every other voice or data codec, even T.38.
The Mediatrix unit supports the MIKEY protocol using pre-shared keys (MIKEY-PS) or the SDES protocol for
negotiating SRTP keys.
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Chapter 34 - Security
Security Parameters
 To set the RTP stream security parameters:
1.
In the web interface, click the Media link, then the Security sub-link.
Figure 163: Media – Security Web Page
2
5
4
6
7
2.
Select to which endpoint (interface) you want to apply the changes in the Select Endpoint dropdown menu at the top of the window.
You have the choice between Default and the interfaces of your Mediatrix unit. The number of
interfaces available vary depending on the Mediatrix unit model you have.
3.
Select whether or not you want to override one or more of the available default security parameters
in the Endpoint Specific drop-down menu.
This menu is available only in the specific endpoints configuration.
4.
In the Security section of the Security page, select the RTP payload mode in the Mode drop-down
menu.
The unit relies on these modes when negotiating an audio stream.
Table 318: Default RTP Mode
Mode
Description
Unsecure
The Mediatrix unit supports only unsecure RTP. It rejects secure RTP
offers it receives.
Secure
The Mediatrix unit supports only secure RTP. It rejects unsecure RTP
offers it receives.
Secure with fallback
The Mediatrix unit supports both secure and unsecure RTP. It prioritizes
secure RTP but permits unsecure RTP fallback when the remote peer
does not support security.
The TLS SIP transport must usually be enabled for secure audio negotiation via SDP (refer to the
Caution box above). See “Chapter 30 - SIP Transport Parameters” on page 271 for more details.
The RTP mode is reflected in the SIP/SDP payload, with a RTP/AVP for unsecure RTP, and a RTP/
SAVP for secure RTP.
The following basic rules apply when sending units capabilities via SDP:
346
•
When the RTP mode is set to Unsecure, the Mediatrix unit offers/answers with only one
active RTP/AVP audio stream. Any other audio stream present in the offer is disabled
in the answer.
•
When the RTP mode is set to Secure, the Mediatrix unit offers/answers with only one
active RTP/SAVP audio stream. Any other audio stream present in the offer is disabled
in the answer.
•
When the RTP mode is set to Secure with fallback, the Mediatrix unit offers one RTP/
AVP and one RTP/SAVP audio streams. The unit answers with only the most secure
stream.
Dgw v2.0 Application
Security Parameters
Software Configuration Guide
•
5.
If the remote unit answers to an offer with both RTP/AVP and RTP/SAVP streams
enabled, a new offer is sent with only RTP/SAVP enabled.
Select the key management protocol for SRTP in the Key Management drop-down menu.
Table 319: Key Management Protocol
Protocol
Description
Mikey
Use MIKEY (Multimedia Internet KEYing).
Sdes
Use SDES (Security DEScriptions).
This parameter has no effect if the Mode parameter is set to Unsecure.
If the unit receives an offer with both MIKEY and SDES, only the configured key management
protocol is kept.
6.
Select the encryption type to be used with SRTP in the Encryption drop-down menu.
Table 320: Default RTP Mode
Encryption
Description
Null
No encryption. It is ignored for the Sdes Key Management as
defined in Step 3. Use only for debug.
AesCm128
AES (Advanced Encryption Standard) Counter Mode 128 bits.
This parameter has no effect if the Mode parameter is set to Unsecure.
7.
Select whether or not to enable T.38 even if the call has been established previously in SRTP in the
Allow Unsecure T.38 with Secure RTP drop-down menu.
Table 321: Default RTP Mode
Mode
Description
Disable
T.38 is disabled for SRTP calls.
Enable
T.38 is enabled for SRTP calls.
Caution: Enabling this parameter opens a security hole,
because T.38 is an unsecure protocol.
This menu is available only in the default configuration.
Note that this parameter has no effect if the Mode parameter is set to Unsecure.
8.
Click Submit if you do not need to set other parameters.
Enforcing Symmetric RTP
This section describes configuration that is available only in the MIB parameters of the Mediatrix unit. You can
configure these parameters as follows:



by using a MIB browser
by using the CLI
by creating a configuration script containing the configuration variables
For each bi-directional RTP streams, you can define whether or not to enforce that incoming RTP packets are
from the same source as the destination of outgoing RTP packets.
Enforcing symmetric RTP may prevent legitimate RTP streams coming from a media server from being
processed, for example: Music and conferencing servers.
Dgw v2.0 Application
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Security Parameters
The following parameters are available:
Table 322: Enforce Symmetric RTP Parameters
Parameter
Description
disable
Accept packets from all sources. This is the default value.
enable
Silently discard incoming RTP packets with source address and port differing from the
destination address and port of outgoing packets.
 To enforce symmetric RTP:
1.
In the miptMIB, set the enforceSymmetricRtpEnable variable with the proper behaviour.
You can also use the following line in the CLI or a configuration script:
mipt.enforceSymmetricRtpEnable="Value"
where Value may be as follows:
Figure 164: Symmetric RTP Values
Value Meaning
348
0
disable
1
enable
Dgw v2.0 Application
Statistics Displayed
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Software Configuration Guide
H A P T E R
35
RTP Statistics Configuration
The Mediatrix unit collects meaningful statistics that can be read via the web interface. This chapter describes
how to read and configure the RTP statistics.
Note that the RTP statistics are also available via SNMP and CLI.
Statistics Displayed
The Mediatrix unit collects two types of statistics:


statistics for the last 10 connections
statistics for the last 10 collection periods
The Connection Statistics section displays the statistics for the last 10 connections. You can use the Display
All button to display more information or the Display Overview button to display less information.
The Connection Period Statistics section displays the statistics for the last 10 periods. The period duration is
defined in the Statistics Configuration section. You can use the Display All button to display more information
or the Display Overview button to display less information.
Figure 165: Telephony – RTP Stats Web Page
The following table describes the statistics available.
Table 323: Statistics Displayed
Statistic
Octets Tx
Dgw v2.0 Application
Connection Statistics
Number of octets transmitted during the
connection.
Collection Period Statistics
Number of octets transmitted during the
collection period. This value is obtained by
cumulating the octets transmitted in all
connections that were active during the
collection period.
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Statistics Displayed
Table 323: Statistics Displayed (Continued)
Statistic
350
Connection Statistics
Collection Period Statistics
Octets Rx
Number of octets received during the
connection.
Number of octets received during the
collection period. This value is obtained by
cumulating the octets received in all
connections that were active during the
collection period.
Packets Tx
Number of packets transmitted during the
connection.
Number of packets transmitted during the
collection period. This value is obtained by
cumulating the packets transmitted in all
connections that were active during the
collection period.
Packets Rx
Number of packets received during the
connection.
Number of packets received during the
collection period. This value is obtained by
cumulating the packets received in all
connections that were active during the
collection period.
Packets Lost
Number of packets lost during the
connection. This value is obtained by
substracting the expected number of
packets based on the sequence number
from the number of packets received.
Number of packets lost during the collection
period. This value is obtained by cumulating
the packets lost in all connections that were
active during the collection period.
Min. Jitter
Minimum interarrival time, in ms, during the
connection. All RTP packets belonging to
the connection and received at the RTP
level are considered in the calculation.
Minimum interarrival time, in ms, during the
collection period. This value is the lowest
interarrival jitter for all connections that were
active during the collection period.
Max. Jitter
Maximum interarrival time, in ms, during the
connection. All RTP packets belonging to
the connection and received at the RTP
level are considered in the calculation.
Maximum interarrival time, in ms, during the
collection period. This value is the highest
interarrival jitter for all connections that were
active during the collection period.
Avg. Jitter
Average interarrival time, in ms, during the
connection. All RTP packets belonging to
the connection and received at the RTP
level are considered in the calculation.
Average interarrival time, in ms, during the
collection period. This value is the weighted
average of the interarrival jitter for all
connections that were active during the
collection period. For each connection, the
total jitter of packets received during the
collection period and the total number of
packets received during the collection period
are used in the weighted average
calculation.
Min. Latency
Minimum latency, in ms, during the
connection. The latency value is computed
as one half of the round-trip time, as
measured through RTCP.
Minimum latency, in ms, during the
collection period. This value is the lowest
latency for all connections that were active
during the collection period.
Max. Latency Maximum latency, in ms, during the
connection. The latency value is computed
as one half of the round-trip time, as
measured through RTCP.
Maximum latency, in ms, during the
collection period. This value is the highest
latency for all connections that were active
during the collection period.
Dgw v2.0 Application
Statistics Configuration
Software Configuration Guide
Table 323: Statistics Displayed (Continued)
Statistic
Avg. Latency
Connection Statistics
Collection Period Statistics
Average latency, in ms, during the
connection. The latency value is computed
as one half of the round-trip time, as
measured through RTCP.
Average latency, in ms, during the collection
period. This value is the weighted average
of the latency for all connections that were
active during the collection period. For each
connection, the total latency of packets
received during the collection period and the
total number of packets received during the
collection period are used in the weighted
average calculation.
Statistics Configuration
You can define how to collect the statistics. The statistics are sent as syslog messages, so you must properly
set the syslog information before setting the statistics. You must set the Media IP Transport (MIPT) service to
the Info or Debug level. See “Syslog Daemon Configuration” on page 35 for more details on how to configure
the Syslog.
 To configure how to collect statistics:
1.
In the web interface, click the Telephony link, then the RTP Stats sub-link.
Figure 166: Telephony – RTP stats Web Page
2
3
4
2.
Set the Collection Period field with the collection period duration in minutes.
Putting a value of 0 disables the collection period statistics feature.
3.
Set the End-of-Connection Notification drop-down menu with the proper behaviour.
Table 324: End-of-Connection Notification
Parameter
Dgw v2.0 Application
Description
Enable
Notifications are generated.
Disable
Notifications are not generated.
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Chapter 35 - RTP Statistics Configuration
4.
Channel Statistics
Set the End-of-Period Notification drop-down menu with the proper behaviour.
Table 325: End-of-Period Notification
Parameter
5.
Description
Enable
Notifications are generated.
Disable
Notifications are not generated.
If you do not need to set other parameters, do one of the following:
•
To save your settings, click Submit.
•
To save your settings and reset the statistics of the current period., click Submit &
Reset Current Collection Period Statistics.
The previous periods are left unchanged.
Channel Statistics
This section describes how to access data available only in the MIB parameters of the Mediatrix unit. You can
display these parameters as follows:


by using a MIB browser
by using the CLI
The channel statistics are cumulated RTP statistics for all calls using a specific channel of a telephony
interface. Statistics are updated at the end of each call.
The statistics are associated to the channel in use at the end of the call. In some cases, such as in hold/resume
scenarios, the channel assignment may change during a call. This can result in discrepancies between the
RTP statistics and the actual usage of the telephony interface.
The following are the channel statistics the Mediatrix unit keeps.
Table 326: Channel Statistics
MIB Variable
352
Statistics Description
PacketsSent
Number of packets transmitted on the channel since
service start. This value is obtained by cumulating the
packets transmitted in all the connections that ended
during the collection period.
PacketsReceived
Number of packets received on the channel since
service start. This value is obtained by cumulating the
packets received in all the connections that ended
during the collection period.
BytesSent
Number of bytes transmitted on the channel since
service start. This value is obtained by cumulating the
bytes transmitted in all the connections that ended
during the collection period.
BytesReceived
Number of bytes received on the channel since service
start. This value is obtained by cumulating the bytes
received in all the connections that ended during the
collection period.
AverageReceiveInterarr
ivalJitter
Average interarrival time, in microseconds, for the
channel since service start. This value is based on the
average interarrival jitter of each call ended during the
collection period. The value is weighted by the duration
of the calls.
Dgw v2.0 Application
Channel Statistics
Software Configuration Guide
 To display channel statistics:
1.
In the miptMIB, go to the ChannelStatistics table.
You can also use the following line in the CLI:
get mipt.channelStatistics
 To reset channel statistics values to zero:
1.
In the miptMIB, set ChannelStatistics.Reset to Reset for the endpoint to reset.
You can also use the following line in the CLI:
set mipt.channelStatistics.Reset=Reset
2.
In the miptMIB, set ChannelStatistics[EpChannelId=channelStatisticsEpChannelId].Reset
to Reset to reset only one specific endpoint.
where:
•
channelStatisticsEpChannelId is the string that identifies the combination of an
endpoint and a channel. The endpoint name is the same as the EpId used to refer to
endpoints in other tables. On endpoints with multiple channels, the channel number
must be appended at the end of the endpoint name, separated with a dash.
You can also use the following line in the CLI:
set mipt.channelStatistics[EpChannelId=channelStatisticsEpChannelId].Reset=Reset
Examples:
Slot3/E1T1-12 refers to endpoint Slot3/E1T1, channel 12.
Phone-Fax1 refers to FXS endpoint Phone-Fax1 on a 4102s.
Port06 refers to FXS endpoint Port06 on 4108/4116/4124.
No channel number is appended to FXS endpoint strings because FXS lines do not support multiple
channels.
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Channel Statistics
Dgw v2.0 Application
Jitter Buffer Configuration
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Software Configuration Guide
H A P T E R
36
Miscellaneous Media Parameters
This chapter describes how to configure parameters that apply to all codecs.




Jitter Buffer Configuration
DTMF Transport Configuration
Machine Detection Configuration
Base Ports Configuration
Jitter Buffer Configuration
The Jitter Buffer section allows you to configure parameters to reduce jitter buffer issues.
 To set the jitter buffer parameters:
1.
In the web interface, click the Media link, then the Misc sub-link.
Figure 167: Media – Misc Web Page
2
3
4
5
6
7
8
9
10
2.
Select to which endpoint (interface) you want to apply the changes in the Select Endpoint dropdown menu at the top of the window.
You have the choice between Default and the interfaces of your Mediatrix unit. The number of
interfaces available vary depending on the Mediatrix unit model you have.
3.
In the Jitter Buffer section, if you have selected a specific endpoint, select whether or not you want
to override the jitter buffer parameters set in the Default configuration in the Endpoint Specific dropdown menu.
This menu is available only in the specific endpoints configuration.
4.
Dgw v2.0 Application
Select the jitter buffer level in the Level drop-down menu.
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Chapter 36 - Miscellaneous Media Parameters
Jitter Buffer Configuration
Jitter is an abrupt and unwanted variation of one or more signal characteristics, such as the interval
between successive pulses or the frequency or phase of successive cycles. An adaptive jitter buffer
usually consists of an elastic buffer in which the signal is temporarily stored and then retransmitted
at a rate based on the average rate of the incoming signal.
Table 327: Jitter Buffer Levels
Level
Description
Optimize Latency The jitter buffer is set to the lowest effective value to minimize the latency.
Voice cut can be heard if the network is not optimal. The predefined values
are as follows:
Normal
Fax / Modem
5.
Minimum value: 10 ms
•
Maximum value: 40 ms
The jitter buffer tries to find a good compromise between the latency and the
voice quality. This setting is recommended in private networks. The
predefined values are as follows:
Optimize Quality
Custom
•
•
Minimum value: 30 ms
•
Maximum value: 90 ms
The jitter buffer is set to a high value to minimize the voice cuts at the cost of
high latency. This setting is recommended in public networks. The
predefined values are as follows:
•
Minimum value: 50 ms
•
Maximum value: 125 ms
The jitter buffer is set to maximum. The Fax/Modem transmission is very
sensitive to voice cuts but not to latency, so the fax has a better chance of
success with a high buffer. The predefined values are as follows:
•
Minimum value: 70 ms
•
Maximum value: 135 ms
The jitter buffer uses the configuration of the Minimum and Maximum
variables (Steps 4 and 5).
If you have selected the Custom level, define the target jitter buffer length in the Minimum field of
the Voice Call part.
The adaptive jitter buffer attempts to hold packets to the minimal holding time. This is the minimal
delay the jitter buffer adds to the system. The minimal jitter buffer is in ms and must be equal to or
smaller than the maximal jitter buffer.
Values range from 0 ms to 135 ms. The default value is 30 ms. You can change values by
increments of 1 ms, but Media5 recommends to use multiples of 5 ms. The minimal jitter buffer
should be a multiple of ptime.
It is best not to set the minimal jitter value below the default value. Setting a minimal jitter buffer
below 5 ms could cause an error. Jitter buffer adaptation behaviour varies from one codec to
another. See “About Changing Jitter Buffer Values” on page 357 for more details.
6.
If you have selected the Custom level, define the maximum jitter buffer length in the Maximum field
of the Voice Call part.
This is the highest delay the jitter buffer is allowed to introduce. The jitter buffer length is in ms and
must be equal to or greater than the minimum jitter buffer.
Values range from 0 ms to 135 ms. The default value is 125 ms. You can change values by
increments of 1 ms, but Media5 recommends to use multiples of 5 ms. The maximal jitter buffer
should be a multiple of ptime.
The maximum jitter buffer value should be equal to the minimum jitter buffer value + 4 times the
ptime value. Let’s say for instance that:
•
356
Minimum jitter buffer value is 30 ms
Dgw v2.0 Application
Jitter Buffer Configuration
Software Configuration Guide
•
Ptime value is 20 ms
The maximum jitter buffer value should be: 30 + 4x20 = 110 ms
7.
If you have selected the Custom level, define the voiceband data custom jitter buffer type in the
Playout Type drop-down menu of the Data Call part.
This is the algorithm to use for managing the jitter buffer during a call. The Nominal field value
serves as the delay at the beginning of the call and might be adapted afterwards based on the
selected algorithm.
Table 328: Voiceband Data Custom Jitter Buffer Type
Level
8.
Description
Adaptive
Immediately
The nominal delay varies based on the estimated packet jitter. Playout
adjustment is done immediately when the actual delay goes out of bounds of
a small window around the moving nominal delay.
Adaptive Silence
The nominal delay varies based on the estimated packet jitter. Playout
adjustment is done based on the actual delay going out of bounds of a small
window around the moving nominal delay. The adjustment is deferred until
silence is detected (either from playout buffer underflow or by analysis of
packet content). Playout adjustment is also done when overflow or
underflow events occur.
Fixed
The nominal delay is fixed to the value of the Nominal field value and does
not change thereafter. Playout adjustment is done when overflow or
underflow events occur.
If you have selected the Custom level, define the voiceband data jitter buffer minimal length (in
milliseconds) in the Minimum field of the Data Call part.
The voiceband data jitter buffer minimal length is the delay the jitter buffer tries to maintain. The
minimal jitter buffer MUST be equal to or smaller than the voiceband data maximal jitter buffer.
The minimal jitter buffer should be a multiple of ptime.
This value is not available when the Playout Type drop-down menu is set to Fixed.
9.
If you have selected the Custom level, define the voiceband data custom jitter buffer nominal length
in the Nominal field of the Data Call part.
The jitter buffer nominal length (in milliseconds) is the delay the jitter buffer uses when a call begins.
The delay then varies depending on the type of jitter buffer.
In adaptive mode, the nominal jitter buffer should be equal to (voice band data minimal jitter buffer
+ voice band data maximal jitter buffer) / 2.
10.
If you have selected the Custom level, define the default voiceband data custom jitter buffer
maximal length in the Maximum field of the Data Call part.
The jitter buffer maximal length (in milliseconds) is the highest delay the jitter buffer is allowed to
introduce. The maximal jitter buffer MUST be equal to or greater than the minimal jitter buffer.
The maximal jitter buffer should be a multiple of ptime.
The maximal jitter buffer should be equal to or greater than voiceband data minimal jitter buffer + (4
* ptime) in adaptive mode.
See “About Changing Jitter Buffer Values” on page 357 for more details.
11.
Click Submit if you do not need to set other parameters.
About Changing Jitter Buffer Values
Media5 recommends to avoid changing the target and maximum jitter buffer values unless experiencing or
strongly expecting one of the following symptoms:


Dgw v2.0 Application
If the voice is scattered, try to increase the maximum jitter buffer value.
If the delay in the voice path (end to end) is too long, you can lower the target jitter value, but
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DTMF Transport Configuration
ONLY if the end-to-end delay measured matches the target jitter value.
For instance, if the target jitter value is 50 ms, the maximum jitter is 300 ms and the delay measured
is 260 ms, it would serve nothing to reduce the target jitter. However, if the target jitter value is
100 ms and the measured delay is between 100 ms and 110 ms, then you can lower the target jitter
from 100 ms to 30 ms.
Starting a Call in Voiceband Data Mode
This section describes configuration that is available only in the MIB parameters of the Mediatrix unit. You can
configure these parameters as follows:



by using a MIB browser
by using the CLI
by creating a configuration script containing the configuration variables
You can define whether or not a call should be started in voiceband data mode.
The following values are available:
Table 329: Voiceband Data Mode Parameters
Parameter
Description
Disable
The call is started in voice mode. A fax/modem tone detection triggers a transition from voice
to voiceband data according to the configuration in the Machine Detection Group
(“Miscellaneous Media Parameters” on page 355).
Enable
The call is started in voiceband data mode.
 To start a call in voiceband data mode:
1.
In the telIfMIB, set the voiceband data mode in the InteropStartCallInVbdEnable variable.
You can also use the following line in the CLI or a configuration script:
telIf.InteropStartCallInVbdEnable="Value"
where Value may be as follows:
Table 330: Voiceband Data Mode Values
Value
Method
0
Disable
1
Enable
DTMF Transport Configuration
The DTMF Transport section allows you to set the DTMF transport parameters of the Mediatrix unit.
 To set DTMF transport parameters:
1.
Select to which endpoint (interface) you want to apply the changes in the Select Endpoint dropdown menu at the top of the window.
You have the choice between Default and the interfaces of your Mediatrix unit. The number of
interfaces available vary depending on the Mediatrix unit model you have.
2.
In the DTMF Transport section of the Misc page, select whether or not you want to override the
DTMF transport parameters set in the Default configuration in the Endpoint Specific drop-down
menu.
This menu is available only in the specific endpoints configuration.
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DTMF Transport Configuration
Software Configuration Guide
Figure 168: DTMF Transport Section
2
4
3.
3
Select the DTMF transport type in the Transport Method drop-down menu.
The following choices are available:
Table 331: DTMF Transport Type Parameters
Transport Parameter
4.
Description
In-band
The DTMFs are transmitted like the voice in the RTP
stream.
Out-of-band using RTP
The DTMFs are transmitted as per RFC 2833. This
parameter also works with SRTP.
Out-of-band using SIP
The DTMFs are transmitted as per draft-choudhuri-sip-infodigit-00.
Signaling protocol Dependant
The signalling protocol has the control to select the DTMF
transport mode. The SDP body includes both RFC 2833
and draft-choudhuri-sip-info-digit-00 in that order of
preference.
If you have selected the Out-of-band using SIP transport method, select the method used to
transport DTMFs out-of-band over the SIP protocol in the SIP Transport Method drop-down menu.
This menu is available only in the default endpoint configuration.
Table 332: DTMF Out-of-Band Transport Methods
Method
Description
draftChoudhuriSipInfoDigit00 Transmits DTMFs by using the method defined in draftchoudhuri-sip-info-digit-00. Only the unsolicited-digit part is
supported.
DTMF out-of-band
Certain compression codecs such as G.723.1 and G.729 effectively distort voice because they lose
information from the incoming voice stream during the compression and decompression phases. For
normal speech this is insignificant and becomes unimportant. In the case of pure tones (such as DTMF)
this distortion means the receiver may no longer recognize the tones. The solution is to send this
information as a separate packet to the other endpoint, which then plays the DTMF sequence back by regenerating the true tones. Such a mechanism is known as out-of-band DTMF. The Mediatrix unit receives
and sends out-of-band DTMFs as per ITU Q.24. DTMFs supported are 0-9, A-D, *, #.
Dgw v2.0 Application
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Chapter 36 - Miscellaneous Media Parameters
DTMF Transport Configuration
Table 332: DTMF Out-of-Band Transport Methods (Continued)
Method
Info DTMF Relay
Description
Transmits DTMFs by using a custom method. This custom
method requires no SDP negotiation and assumes that the other
peer uses the same method.
It uses a SIP INFO message with a content of type application/
dtmf-relay. The body of the message contains the DTMF
transmitted and the duration of the DTMF:
Signal= 1
Duration= 160
When transmitting, the duration is the one set in the
interopDtmfTransportDuration variable (see “DTMF
Transport over the SIP Protocol” on page 360).
When receiving, the duration of the DTMF received is not used
and the DTMF is played for 100 ms.
DTMFs are transmitted one at a time.
Available digits are “0123456789ABCD*#”. The Mediatrix unit
also supports the “,;p” characters when receiving DTMFs.
5.
If you have selected the Out-of-band using RTP transport method, set the payload type in the
Payload Type field.
You can determine the actual RTP dynamic payload type used for the “telephone-event” in an initial
offer. The payload types available are as per RFC 1890. Available values range from 96 to 127.
6.
Click Submit if you do not need to set other parameters.
DTMF Transport over the SIP Protocol
This section describes configuration that is available only in the MIB parameters of the Mediatrix unit. You can
configure these parameters as follows:



by using a MIB browser
by using the CLI
by creating a configuration script containing the configuration variables
You can set the DTMF duration sent in the INFO message when using the Info DTMF Relay method to
transmit DTMFs (see “Miscellaneous Media Parameters” on page 355, Step 8 for more details).
 To set the DTMF duration sent in the INFO message:
1.
In the sipEpMIB, set the DTMF duration sent in the INFO message when using the infoDtmfRelay
method to transmit DTMFs in the interopDtmfTransportDuration variable.
You can also use the following line in the CLI or a configuration script:
sipEp.interopDtmfTransportDuration="Value"
This value is expressed in milliseconds (ms). The default value is 100 ms.
DTMF Detection
This section describes configuration that is available only in the MIB parameters of the Mediatrix unit. You can
configure these parameters as follows:



by using a MIB browser
by using the CLI
by creating a configuration script containing the configuration variables
The default DTMF detection parameters of the Mediatrix unit may sometimes not be enough to properly detect
the DTMFs. This section describes how to set additional DTMF detection parameters.
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Software Configuration Guide
DTMF Frequencies
The DTMF keypad is laid out in a 4x4 matrix, with each row representing a low frequency, and each column
representing a high frequency. For example, pressing a single key (such as '1') sends a sinusoidal tone of the
two frequencies (697 Hz and 1209 Hz). When the unit is configured to send DTMFs out-of-band, its DSP
detects these DTMFs, removes them from the RTP stream, and sends them out-of-band.
Table 333: DTMF Keypad Frequencies
Low/High (Hz)
1209
1336
1477
1633
697
1
2
3
A
770
4
5
6
B
852
7
8
9
C
941
*
0
#
D
DTMF Detection Configuration
Below is a frequency spectrum analysis of a DTMF (9) with the Frequency in Hertz on the x axis and the Power
in dBm on the y axis. The low and high frequencies of the DTMF are in red and you can clearly see that they
are the most powerful frequencies in the signal.
Figure 169: DTMF Detection Example
 To configure the DTMF detection:
1.
In the telIfMIB, define how the Rise Time criteria should be configured for DTMF detection in the
interopDtmfDetectionRiseTimeCriteria variable.
You can also use the following line in the CLI or a configuration script:
TelIf.InteropDtmfDetection[InterfaceId=xxx].RiseTimeCriteria = "Value"
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where Value may be as follows:
Table 334: DTMF Detection Values
Value
100
Method
CheckSr
Enables the Step Rise criteria and disables the Confirm DTMF SNR
criteria.
The Step Rise criteria compares the current frame energy to the high
frequency power of the previous frame. If the current frame energy is high
enough, then it passes the test, further validating the DTMF.
Disabling the Step Rise criteria may result in deteriorated talk-off
performance, but increases the detection of malformed DTMF.
200
ConfirmSnr Enable the Confirm DTMF SNR criteria and disable the Step Rise criteria.
The Confirm DTMF SNR criteria is an additional Signal-to-noise ratio test
performed before a confirmed DTMF report is sent to finally validate the
DTMF.
2.
Set the interopDtmfDetectionPositiveTwist variable.
You can also use the following line in the CLI or a configuration script:
TelIf.InteropDtmfDetection[InterfaceId=xxx].PositiveTwist = "Value"
When the high-group frequency of a DTMF is more powerful than the low-group frequency, the
difference between the high-group frequency absolute power and the low-group frequency absolute
power must be smaller than or equal to the value set in this variable. Otherwise, the DTMF is not
detected.
Raising this value increases the sensitivity of DTMF detection. Raising this value too high may also
cause false detections of DTMFs.
3.
Set the interopDtmfDetectionNegativeTwist variable.
You can also use the following line in the CLI or a configuration script:
TelIf.InteropDtmfDetection[InterfaceId=xxx].NegativeTwist = "Value"
Defines the value for the Negative Twist DTMF detection parameter.
When the low-group frequency of a DTMF is more powerful than the high-group frequency, the
difference between the low-group frequency absolute power and the high-group frequency absolute
power must be smaller than or equal to the value set in this parameter. Otherwise, the DTMF is not
detected.
Raising this value increases the sensitivity of DTMF detection. Raising this value too high may also
cause false detections of DTMFs.
4.
Set the interopDtmfDetectionMaxPowerThreshold variable.
You can also use the following line in the CLI or a configuration script:
TelIf.InteropDtmfDetection[InterfaceId=xxx].MaxPowerThreshold = "Value"
The average power of a DTMF must be below the value set in this parameter to be no longer
detected.
The value is expressed in dBm (relative to 1mW of power).
5.
Set the interopDtmfDetectionMinPowerThreshold variable.
You can also use the following line in the CLI or a configuration script:
TelIf.InteropDtmfDetection[InterfaceId=xxx].MinPowerThreshold = "Value"
The average power of a DTMF must be above the value set in this parameter for at least 30ms to
be detected.
The value is expressed in dBm (relative to 1mW of power).
6.
362
Set the interopDtmfDetectionBreakPowerThreshold variable.
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DTMF Transport Configuration
Software Configuration Guide
You can also use the following line in the CLI or a configuration script:
TelIf.InteropDtmfDetection[InterfaceId=xxx].BreakPowerThreshold = "Value"
When the average power of a DTMF falls below the value set in this parameter for at least 20ms, it
is considered that the DTMF ended.
The value is expressed in dBm (relative to 1mW of power).
Using the Payload Type Found in the Answer
This section describes configuration that is available only in the MIB parameters of the Mediatrix unit. You can
configure these parameters as follows:



by using a MIB browser
by using the CLI
by creating a configuration script containing the configuration variables
The default behaviour when sending an initial offer that contains an RFC 2833 payload type is to keep using
that payload type even if the response comes back with a different one. You can set the Mediatrix unit to rather
use the payload type found in the answer.
This feature is effective only if the Transport Method drop-down menu is set to Out-of-band using RTP (see
“Miscellaneous Media Parameters” on page 355 for more details).
The following parameters are available:
Table 335: Payload Type in Answer
Parameter
Description
disable
Keep using the initial payload type. This is the default value.
enable
Use the RFC 2833 payload type found in the received answer.
 To use the payload type found in the answer:
1.
In the sipEpMIB, set the interopUseDtmfPayloadTypeFoundInAnswer variable with the proper
behaviour.
You can also use the following line in the CLI or a configuration script:
sipEp.interopUseDtmfPayloadTypeFoundInAnswer="Value"
where Value may be as follows:
Figure 170: Payload Type Values
Value Meaning
0
disable
1
enable
Quantity of initial packets sent to transmit a DTMF Out-of-Band using RTP
This section describes configuration that is available only in the MIB parameters of the Mediatrix unit. You can
configure these parameters as follows:



by using a MIB browser
by using the CLI
by creating a configuration script containing the configuration variables
You can specify the quantity of packets sent at the beginning of an Out-of-Band DTMF using RTP. This
variable also specifies the quantity of terminating packets that are sent at the end of the DTMF transmission.
Note that this variable has an effect only if the Transport Method drop-down menu is set to Out-of-band
using RTP (see “Miscellaneous Media Parameters” on page 355 for more details).
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Machine Detection Configuration
 To set the initial quantity of RTP packets:
1.
In the miptMIB, set the InteropDtmfRtpInitialPacketQty variable with the proper quantity.
You can also use the following line in the CLI or a configuration script:
mipt.interopDtmfRtpInitialPacketQty="Value"
where Value may be between 1 and 3.
Machine Detection Configuration
The Machine Detection section allows you to set the tone detection parameters of the Mediatrix unit.
 To set Machine detection parameters:
1.
Select to which endpoint (interface) you want to apply the changes in the Select Endpoint dropdown menu at the top of the window.
You have the choice between Default and the interfaces of your Mediatrix unit. The number of
interfaces available vary depending on the Mediatrix unit model you have.
2.
In the Machine Detection section of the Misc page, select whether or not you want to override the
machine detection parameters set in the Default configuration in the Endpoint Specific drop-down
menu.
This menu is available only in the specific endpoints configuration.
Figure 171: Machine Detection Section
2
4
3
5
6
3.
Select whether or not you want to enable fax calling tone (CNG tone) detection in the CNG Tone
Detection drop-down menu.
Table 336: CNG Tone Detection Settings
Setting
Enable
Description
Upon recognition of the CNG tone, the unit switches the communication from
voice mode to fax mode and the CNG is transferred by using the preferred fax
codec.
Note: This option allows for quicker fax detection, but it also increases the risk
of false detection.
Disable
The CNG tone does not trigger a transition from voice to data and the CNG is
transferred in the voice channel.
Note: With this option, faxes are detected later, but the risk of false detection
is reduced.
4.
Select whether or not you want to enable CED tone detection in the CED Tone Detection drop-down
menu.
Table 337: CNG CED Detection Settings
Setting
Enable
364
Description
Upon recognition of the CED tone, the unit behaves as defined in the
Behavior on CED Tone Detection parameter Step 6).
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Base Ports Configuration
Software Configuration Guide
Table 337: CNG CED Detection Settings (Continued)
Setting
Disable
5.
Description
The CED tone does not trigger a transition to fax or voiceband data mode.
The CED is transferred in the voice channel.
Select whether or not you want to enable fax V.21 modulation detection in the V.21 Modulation
Detection drop-down menu.
Table 338: V.21 Modulation Detection Settings
Setting
6.
Description
Enable
Upon recognition of the V.21 modulation tone, the unit switches the
communication from voice mode to fax mode and the signal is transferred by
using the preferred fax codec.
Disable
The V.21 modulation does not trigger a transition from voice to data and the
signal is transferred in the voice channel.
Define the behaviour of the unit upon detection of a CED tone in the Behavior on CED Tone
Detection drop-down menu.
Table 339: CED Tone Detection Settings
Setting
Description
Passthrough
The CED tone triggers a transition from voice to voice band data and is
transferred in the voice channel.. Use this setting when any kind of analog
device (i.e.: telephone, fax or modem) can be connected to this port.
Fax Mode
Upon detection of a CED tone, the unit switches the communication from
voice mode to fax mode and the CED is transferred by using the preferred fax
codec. Only a fax can then be connected to this port.
Note: This parameter has no effect if the CED Tone Detection parameter is set to Disabled.
7.
Click Submit if you do not need to set other parameters.
Base Ports Configuration
The Base Ports section allows you to set the ports that the Mediatrix unit uses for different transports.
This section is available only in the default endpoint configuration.
 To set base ports parameters:
1.
Select to which endpoint (interface) you want to apply the changes in the Select Endpoint dropdown menu at the top of the window.
You have the choice between Default and the interfaces of your Mediatrix unit. The number of
interfaces available vary depending on the Mediatrix unit model you have.
2.
In the Base Ports section of the Misc page, set the UDP port number you want to use as RTP/RTCP
base port in the RTP field.
The RTP/RTCP ports are allocated starting from this base port.
RTP ports number are even and RTCP ports number are odd.
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Base Ports Configuration
The default RTP/RTCP base port is 5004. For instance, assuming that the base port is defined on
5004, if there is currently no ongoing call and there is an incoming or outgoing call, the unit uses the
RTP/RTCP ports 5004 and 5005.
Figure 172: Base Ports Section
2
4
3.
3
Set the UDP port number you want to use as SRTP/SRTCP base port in the SRTP field.
The SRTP/SRTCP ports are allocated starting from this base port.
SRTP ports number are even and SRTCP ports number are odd.
The default SRTP/SRTCP base port is 5004. For instance, assuming that the base port is defined
on 5004, if there is currently no ongoing call and there is an incoming or outgoing call, the unit uses
the SRTP/SRTCP ports 5004 and 5005.
Using the same base port for RTP/RTCP and SRTP/SRTCP does not conflict.
Note that if the media transport is set to “Secure with fallback” (“Chapter 34 - Security” on page 345),
both RTP and SRTP base ports are used at the same time when initiating an outgoing call. If there
is currently no call and the default base ports are used, the RTP port is 5004 and the SRTP port is
the next available port starting from the base port, which is 5006.
4.
Set the port number you want to use as T.38 base port in the T.38 field.
The T.38 ports are allocated starting from this base port.
The default T.38 base port is 6004. For instance, assuming that the base port is defined on 6004 if
there is currently no ongoing call and there is an incoming or outgoing call, the unit uses the T.38
port 6005.
This menu is available only in the default endpoint configuration.
5.
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Click Submit if you do not need to set other parameters.
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Telephony Parameters
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Introduction
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H A P T E R
37
DTMF Maps Configuration
This chapter describes how to configure and use the DTMF maps of the Mediatrix unit.




DTMF maps syntax.
General DTMF maps parameters.
Allowed DTMF maps parameters.
Refused DTMF maps parameters.
Introduction
A DTMF map (also called digit map or dial map) allows you to compare the number users just dialed to a string
of arguments. If they match, users can make the call. If not, users cannot make the call and get an error signal.
It is thus essential to define very precisely a DTMF map before actually implementing it, or your users may
encounter calling problems.
Because the Mediatrix unit cannot predict how many digits it needs to accumulate before transmission, you
could use the DTMF map, for instance, to determine exactly when there are enough digits entered from the
user to place a call.
Syntax
The permitted DTMF map syntax is taken from the core MGCP specification, RFC 2705, section 3.4:
DigitMap = DigitString / '(' DigitStringList ')'
DigitStringList = DigitString 0*( '|' DigitString )
DigitString = 1*(DigitStringElement)
DigitStringElement = DigitPosition ['.']
DigitPosition = DigitMapLetter / DigitMapRange
DigitMapLetter = DIGIT / '#' / '*' / 'A' / 'B' / 'C' / 'D' / 'T'
DigitMapRange = 'x' / '[' 1*DigitLetter ']'
DigitLetter ::= *((DIGIT '-' DIGIT ) / DigitMapLetter)
Where “x” means “any digit” and “.” means “any number of”.
For instance, using the telephone on your desk, you can dial the following numbers:
Table 340: Number Examples
Number
Description
0
Local operator
00
Long distance operator
xxxx
Local extension number
8xxxxxxx
Local number
#xxxxxxx
Shortcut to local number at other corporate sites
91xxxxxxxxxx
Long distance numbers
9011 + up to 15 digits
International number
The solution to this problem is to load the Mediatrix unit with a DTMF map that corresponds to the dial plan.
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Chapter 37 - DTMF Maps Configuration
Syntax
A Mediatrix unit that detects digits or timers applies the current dial string to the DTMF map, attempting a
match to each regular expression in the DTMF map in lexical order.

If the result is under-qualified (partially matches at least one entry in the DTMF map), waits for
more digits.


If the result matches, dials the number.
If the result is over-qualified (i.e., no further digits could possibly produce a match), sends a fast
busy signal.
Special Characters
DTMF maps use specific characters and digits in a particular syntax.
Table 341: DTMF Map Characters
Character
Use
Digits (0, 1, 2... 9) Indicates specific digits in a telephone number expression.
T
The Timer indicates that if users have not dialed a digit for the time defined, it is likely
that they have finished dialing and the SIP Server can make the call.
x
Matches any digit, excluding “#” and “*”.
|
Indicates a choice of matching expressions (OR).
.
Matches an arbitrary number of occurrences of the preceding digit, including 0.
[
Indicates the start of a range of characters.
]
Indicates the end of a range of characters.
How to Use a DTMF Map
Let’s say you are in an office and you want to call a co-worker’s 3-digits extension. You could build a DTMF
map that says “after the user has entered 3 digits, make the call”. The DTMF map could look as follows:
xxx
You could refine this DTMF map by including a range of digits. For instance, you know that all extensions in
your company either begin with 2, 3, or 4. The corresponding DTMF map could look as follows:
[2-4]xx
If the number you dial begins with anything other than 2, 3, or 4, the call is not placed and you get a busy signal.
Combining Several Expressions
You can combine two or more expressions in the same DTMF map by using the “|” operator, which is equal to
OR.
Let’s say you want to specify a choice: the DTMF map is to check if the number is internal (extension), or
external (a local call). Assuming that you must first dial “9” to make an external call, you could define a DTMF
map as follows:
([2-4]xx|9[2-9]xxxxxx)
The DTMF map checks if:


the number begins with 2, 3, or 4 and
the number has 3 digits
If not, it checks if:



the number begins with 9 and
the second digit is any digit between 2 and 9 and
the number has 7 digits
Note: Enclose the DTMF map in parenthesis when using the “|” option.
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Software Configuration Guide
Using the # and * Characters
It may sometimes be required that users dial the “#” or “*” to make calls. This can be easily incorporated in a
DTMF map:
xxxxxxx#
xxxxxxx*
The “#” or “*” character could indicate users must dial the “#” or “*” character at the end of their number to
indicate it is complete. You can specify to remove the “#” or “*” found at the end of a dialed number. See
“General DTMF Maps Parameters” on page 372.
Using the Timer
The Timer indicates that if users have not dialed a digit for the time defined, it is likely that they have finished
dialing and the Mediatrix unit can make the call. A DTMF map for this could be:
[2-9]xxxxxxT
Note: When making the actual call and dialing the number, the Mediatrix unit automatically removes the “T”
found at the end of a dialed number, if there is one (after a match). This character is for indication purposes
only.
See “General DTMF Maps Parameters” on page 372 for more details.
Calls Outside the Country
If your users are making calls outside their country, it may sometimes be hard to determine exactly the number
of digits they must enter. You could devise a DTMF map that takes this problem into account:
001x.T
In this example, the DTMF map looks for a number that begins with 001, and then any number of digits after
that (x.).
Example
Table 340 on page 369 outlined various call types one could make. All these possibilities could be covered in
one DTMF map:
(0T|00T|[1-7]xxx|8xxxxxxx|#xxxxxxx|91xxxxxxxxxx|9011x.T)
Validating a DTMF Map
The Mediatrix unit validates the DTMF map as you are entering it and it forbids any invalid value.
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General DTMF Maps Parameters
General DTMF Maps Parameters
The following are the general DTMF maps parameters you can set.
 To set the general DTMF map parameters:
1.
In the web interface, click the Telephony link, then the DTMF Maps sub-link.
Figure 173: Telephony – DTMF Maps Web Page
2
2.
3
4
5
In the General Configuration section, define the time, in milliseconds (ms), between the start of the
dial tone and the receiver off-hook tone, if no DTMF is detected, in the First DTMF Timeout field.
Values range from 1000 ms to 180000 ms. The default value is 20000 ms.
If you want to set a different First DTMF Timeout value for one or more endpoints, click the Edit
Endpoints button (see “Configuring Timeouts per Endpoint” on page 373 for more details).
3.
Define the value, in milliseconds (ms), of the “T” digit in the Inter Digit Timeout field.
The “T” digit expresses a time lapse between the detection of two DTMFs. Values range from 500
ms to 10000 ms. The default value is 3000 ms.
If you want to set a different Inter Digit Timeout value for one or more endpoints, click the Edit
Endpoints button (see “Configuring Timeouts per Endpoint” on page 373 for more details).
4.
Define the total time, in milliseconds (ms), the user has to dial the DTMF sequence in the
Completion Timeout field.
The timer starts when the dial tone is played. When the timer expires, the receiver off-hook tone is
played. Values range from 1000 ms to 180000 ms. The default value is 60000 ms.
If you want to set a different Completion Timeout value for one or more endpoints, click the Edit
Endpoints button (see “Configuring Timeouts per Endpoint” on page 373 for more details).
5.
In the DTMF Maps Digit Detection (FXO/FXS) drop-down menu, define when a digit is processed
through the DTMF maps.
This parameters is available only when the unit has FXS or FXO ports.
Table 342: DTMF Maps Digit Detection Parameters
Parameter
6.
372
Description
When
Pressed
Digits are processed as soon as they are pressed. This can lead to a digit leak in
the RTP at the beginning of a call if the voice stream is established before the last
digit is released.
When
Released
Digits are processed only when released. This option increases the delay needed
to match a dialed string to a DTMF map. There is also an impact on the First DTMF
Timeout, Inter Digit Timeout and Completion Timeout parameters since the timers
are stopped at the end of a digit instead of the beginning.
Click Submit if you do not need to set other parameters.
Dgw v2.0 Application
Allowed DTMF Maps
Software Configuration Guide
Configuring Timeouts per Endpoint
You can set a different timeout value for one or more endpoints.
 To set a different value per endpoint:
1.
In the General Configuration section of the DTMF Maps page, click the Edit Endpoints button.
The following window is displayed:
Figure 174: DTMF Map Timeout Section
2.
Set the Override drop-down menu for the endpoint you want to set to Enable.
3.
Change the value of one or more timeouts as required.
4.
Repeat for each endpoint that you want to modify.
5.
Click Submit when finished.
Allowed DTMF Maps
You can create/edit ten DTMF maps for the Mediatrix unit. DTMF map rules are checked sequentially. If a
telephone number potentially matches two of the rules, the first rule encountered is applied.
 To set up DTMF maps:
1.
In the DTMF Map drop-down menu at the top of the window, select Allowed.
The Allowed DTMF Map section displays.
2.
In the Allowed DTMF Map section – Enable column, enable one or more DTMF maps by selecting
the corresponding Enable choice.
Figure 175: Allowed DTMF Map Section
2
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4
5
6
7
8
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Chapter 37 - DTMF Maps Configuration
3.
Allowed DTMF Maps
Select the entity to which apply the allowed DTMF map in the Apply to column.
Table 343: DTMF Map Entity
Parameter
4.
Description
Unit
The DTMF map entry applies to the unit.
Endpoint
The DTMF map applies to a specific endpoint. The endpoint is specified in the
Endpoint column of the same row.
Enter a string that identifies an endpoint in other tables in the Endpoint column.
This field is available only if you have selected the Endpoint entity in the previous step for the
specific row.
You can specify more than one endpoint. In that case, the endpoints are separated with a comma
(,). You can use the Suggestions column’s drop-down menu to select between suggested values, if
any.
5.
Define the DTMF map string that is considered valid when dialed in the DTMF Map column.
The string must use the syntax described in “DTMF Maps Configuration” on page 369. A DTMF map
string may have a maximum of 64 characters.
6.
Enter the DTMF transformation to apply to the signalled DTMFs before using it as call destination
in the Transformation column.
The following are the rules you must follow; “x” represents the signalled number.
•
Add before “x” the DTMF to prefix or/and after “x” the suffix to add. Characters
“0123456789*# ABCD” are allowed.
•
Use a sequence of DTMFs between “{}” to remove a prefix/suffix from the dialed
number if present. Use before “x” to remove a prefix and after “x” to remove a suffix.
Characters “0123456789*#ABCD” are allowed.
•
Use a number between “()” to remove a number of DTMFs. Use before “x” to remove
DTMFs at the beginning of the number and after “x” to remove DTMFs at the end.
Characters “0123456789” are allowed.
The transformations are applied in order from left to right.
The following table gives an example with “18195551111#” as signalled number.
Table 344: DTMF Map Transformation Examples
Action
7.
Transformation
Result
Add the prefix “0” to the dialed number
0x
018195551111#
Remove the suffix “#” from the dialed
number
x{#}
18195551111
Remove the first four DTMFs from the dialed
number
(4)x
5551111#
Remove the international code and
termination and replace the area code by
another one
(1){819}514x{#}
5145551111
Replace the signalled DTMFs by “3332222”
3332222
3332222
Define the target to use when the DTMF map matches in the Target column.
This allows associating a target (FQDN) with a DTMF map. This defines a destination address to
use when the DTMF map matches. This address is used as destination for the INVITEs in place of
the “home domain proxy”. This is useful for such features as the speed dial and emergency call.
The default target is used when the value is empty.
The dialed DTMFs are not used if the target contains a user name.
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Refused DTMF Maps
Software Configuration Guide
8.
Enable/Disable the emergency process of the call in the Emergency column.
•
Disable: The call is processed normally.
•
Enable: The call is processed as emergency.
The Emergency Call service (also called urgent gateway) allows a “911”-style service. It allows a
user to dial a special DTMF map resulting in a message being sent to a specified urgent gateway,
bypassing any other intermediaries.
If enabled, whenever the user dials the specified DTMF map, a message is sent to the target
address.
9.
Click Submit if you do not need to set other parameters.
Refused DTMF Maps
A refused DTMF map forbids to call specific numbers; for instance, you want to accept all 1-8xx numbers
except 1-801. You can create/edit ten refused DTMF maps for the Mediatrix unit.
A refused DTMF map applies before an allowed DTMF map.
 To set up refused DTMF maps:
1.
In the DTMF Map drop-down menu at the top of the window, select Refused.
The Refused DTMF Map section displays.
2.
In the Refused DTMF Map section – Enable column, enable one or more DTMF maps by selecting
the corresponding Enable choice.
Figure 176: Refused DTMF Map Section
2
3.
3
4
5
Select the entity to which apply the refused DTMF map in the Apply to column.
Table 345: DTMF Map Entity
Parameter
4.
Description
Unit
The DTMF map entry applies to the unit.
Endpoint
The DTMF map applies to a specific endpoint. The endpoint is specified in the
Endpoint column of the same row.
Enter a string that identifies an endpoint in other tables in the Endpoint column.
This field is available only if you have selected the Endpoint entity in the previous step for the
specific row.
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Refused DTMF Maps
You can specify more than one endpoint. In that case, the endpoints are separated with a comma
(,). You can use the Suggestions column’s drop-down menu to select between suggested values, if
any.
5.
Define the DTMF map string that is considered valid when dialed in the DTMF Map column.
The string must use the syntax described in “DTMF Maps Configuration” on page 369. A DTMF map
string may have a maximum of 64 characters.
6.
376
Click Submit if you do not need to set other parameters.
Dgw v2.0 Application
Call Forward On Busy
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H A P T E R
38
Call Forward Configuration
This chapter describes how to set three types of Call Forward:



On Busy
On No Answer
Unconditional
You can use two types of configuration:


Default configurations that apply to all the endpoints of the Mediatrix unit.
Specific configurations that override the default configurations. You can define specific
configurations for each endpoint in your Mediatrix unit.
Note: This web page is available only on the following models:
• Mediatrix 3208 / 3216
• Mediatrix 3308 / 3316
• Mediatrix 3716
• Mediatrix 3731
• Mediatrix 3732
• Mediatrix 3741
• Mediatrix 3742
• Mediatrix 4100 Series
• Mediatrix LP Series
• Mediatrix C7 Series
Call Forward On Busy
You can automatically forward the incoming calls of your users to a pre-determined target if they are already
on the line. The user does not have any feedback that a call was forwarded.
You can enable the Call Forward On Busy feature in two ways:
Dgw v2.0 Application

By allowing the user to configure the call forward activation and its destination via the handset
(Steps 4-6).

By manually enabling the service (Steps 7-8).
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Chapter 38 - Call Forward Configuration
Call Forward On Busy
 To set the Call Forward On Busy feature:
1.
In the web interface, click the Telephony link, then the Call Forward sub-link.
Figure 177: Telephony – Call Forward Web Page
2
3
4
5
7
6
8
2.
Select to which endpoint you want to apply the changes in the Select Endpoint drop-down menu at
the top of the window.
You have the choice between Default and all FXS endpoints your Mediatrix unit has.
3.
In the Call Forward On Busy section, define whether or not you want to override the Call Forward
On Busy parameters set in the Default configuration in the Endpoint Specific drop-down menu.
This menu is available only in the specific endpoints configuration.
4.
Enable the Call forward configuration via handset service by setting the Allow Activation via
Handset drop-down menu to Enable.
You also need to configure the activation and deactivation DTMF maps (steps 5 and 6).
If you select Disable, this does not disable the call forward, but prevents the user from activating or
deactivating the call forward service. The user will not be able to use the digits used to activate and
deactivate the call forward service.
5.
Define the digits that users must dial to start the service in the DTMF Map Activation field.
This field is available only in the Default configuration.
For instance, you could decide to put “*72” as the sequence to activate the service. This sequence
must be unique and follow the syntax for DTMF maps (see “Chapter 37 - DTMF Maps Configuration”
on page 369). Dialing this DTMF map does not have any effect unless the service’s status is
“enabled”.
The activating sequence is set for all the endpoints of the Mediatrix unit. You cannot have a different
sequence for each endpoint.
6.
Define the digits that users must dial to stop the service in the DTMF Map Deactivation field.
This field is available only in the Default configuration.
For instance, you could decide to put “*73” as the sequence to deactivate the service. This
sequence must be unique and follow the syntax for DTMF maps (see “Chapter 37 - DTMF Maps
Configuration” on page 369). Dialing this DTMF map does not have any effect unless the service’s
status is “enabled”.
The deactivating sequence is set for all the endpoints of the Mediatrix unit. You cannot have a
different sequence for each endpoint.
7.
Set the call forward service in the Activation field to Inactive or Active.
Table 346: Activation State
State
Inactive
378
Description
The call forward service is not available on the telephone connected to the specific
endpoint. A call to this endpoint is not forwarded if the endpoint is busy.
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Table 346: Activation State (Continued)
State
Active
Description
The call forward service is available on the telephone connected to the specific
endpoint. A call to the endpoint is forwarded to the specified destination if the
endpoint is busy. You must define the call forward destination in the Forwarding
Address field (Step 8). The call forward service behaves as if it is inactive if the
Forwarding Address is empty.
To let the user activate or deactivate this service with his or her handset, see steps 4, 5, and 6. In
that case, the field is automatically updated to reflect the activation status.
8.
Define the address to which forward incoming calls in the Forwarding Address field.
Accepted formats are:
•
telephone numbers (5551111)
•
SIP URLs such as ”scheme:user@host”. For instance, “sip:user@foo.com”.
This string is used literally, so cosmetic symbols (such as the dash in “555-xxxx”) should not be
present.
9.
Click Submit if you do not need to set other parameters.
Configuring Call Forward on Busy via Handset
The following is the procedure to use this service on the user’s telephone.
 To forward calls:
1.
Take the receiver off-hook.
2.
Wait for the dial tone.
3.
Dial the sequence implemented to activate the call forward on busy service.
This sequence could be something like *72.
4.
Wait for the stutter dial tone (three “beeps”) followed by the dial tone.
5.
Dial the number to which you want to forward your calls. Dial any access code if required.
6.
Wait for three “beeps” followed by a silent pause.
The call forward is established.
7.
Hang up your telephone.
 To cancel the call forward:
1.
Take the receiver off-hook.
2.
Wait for the dial tone.
3.
Dial the sequence implemented to deactivate the call forward on busy service.
This sequence could be something like *73.
4.
Wait for the transfer tone (three “beeps”) followed by the dial tone.
The call forward is cancelled.
5.
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Hang up your telephone.
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Call Forward On No Answer
Call Forward On No Answer
You can forward the incoming calls of your users to a pre-determined target if they do not answer their
telephone before a specific amount of time. The user does not have any feedback that a call was forwarded.
You can enable the Call Forward On Busy feature in two ways:

By allowing the user to configure the call forward activation and its destination via the handset
(Steps 3-5).

By manually enabling the service (Steps 6-8).
 To set the Call Forward On No Answer feature:
1.
Select to which endpoint you want to apply the changes in the Select Endpoint drop-down menu at
the top of the window.
You have the choice between Default and all FXS endpoints your Mediatrix unit has.
2.
In the Call Forward On No Answer section, define whether or not you want to override the Call
Forward On No Answer parameters set in the Default configuration in the Endpoint Specific dropdown menu.
This menu is available only in the specific endpoints configuration.
Figure 178: Telephony – Call Forward on No Answer section
2
3
4
6
8
3.
5
7
Enable the Call forward configuration via handset service by setting the Allow Activation via
Handset drop-down menu to Enable.
You also need to configure the activation and deactivation DTMF maps (steps 4 and 5).
If you select Disable, this does not disable the call forward, but prevents the user from activating or
deactivating the call forward service. The user will not be able to use the digits used to activate and
deactivate the call forward service.
4.
Define the digits that users must dial to start the service in the DTMF Map Activation field.
This field is available only in the Default configuration.
For instance, you could decide to put “*74” as the sequence to activate the service. This sequence
must be unique and follow the syntax for DTMF maps (see “Chapter 37 - DTMF Maps Configuration”
on page 369). Dialing this DTMF map does not have any effect unless the service’s status is
“enabled”.
The activating sequence is set for all the endpoints of the Mediatrix unit. You cannot have a different
sequence for each endpoint.
5.
Define the digits that users must dial to stop the service in the DTMF Map Deactivation field.
This field is available only in the Default configuration.
For instance, you could decide to put “*75” as the sequence to deactivate the service. This
sequence must be unique and follow the syntax for DTMF maps (see “Chapter 37 - DTMF Maps
Configuration” on page 369). Dialing this DTMF map does not have any effect unless the service’s
status is “enabled”.
The deactivating sequence is set for all the endpoints of the Mediatrix unit. You cannot have a
different sequence for each endpoint.
6.
380
Define the time, in milliseconds, the telephone keeps ringing before the call forwarding activates in
the Timeout field.
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7.
Set the status of the service in the Activation field to Inactive or Active.
Table 347: Activation State
State
Description
Inactive
The call forward service is not available on the telephone connected to the specific
endpoint. A call to this endpoint is not forwarded if the endpoint is busy.
Active
The call forward service is available on the telephone connected to the specific
endpoint. A call to the endpoint is forwarded to the specified destination if the
endpoint is busy. You must define the call forward destination in the Forwarding
Address field (Step 8). The call forward service behaves as if it is inactive if the
Forwarding Address is empty.
To let the user activate or deactivate this service with his or her handset, see steps 3, 4, and 5. In
that case, the field is automatically updated to reflect the activation status.
8.
Define the address to which forward incoming calls in the Forwarding Address field.
Accepted formats are:
•
telephone numbers (5551111)
•
SIP URLs such as ”scheme:user@host”. For instance, “sip:user@foo.com”.
This string is used literally, so cosmetic symbols (such as the dash in “555-xxxx”) should not be
present.
9.
Click Submit if you do not need to set other parameters.
Configuring Call Forward on Answer via Handset
The following is the procedure to use this service on the user’s telephone.
 To forward calls:
1.
Take the receiver off-hook.
2.
Wait for the dial tone.
3.
Dial the sequence implemented to activate the call forward on no answer service.
This sequence could be something like *74.
4.
Wait for the transfer tone (three “beeps”) followed by the dial tone.
5.
Dial the number to which you want to forward your calls. Dial any access code if required.
6.
Wait for three “beeps” followed by a silent pause.
The call forward is established.
7.
Hang up your telephone.
 To cancel the call forward:
1.
Take the receiver off-hook.
2.
Wait for the dial tone.
3.
Dial the sequence implemented to deactivate the call forward on no answer service.
This sequence could be something like *75.
4.
Wait for the stutter dial tone (three “beeps”) followed by the dial tone.
The call forward is cancelled.
5.
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Hang up your telephone.
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Call Forward Unconditional
Call Forward Unconditional
The Call Forward Unconditional feature allows users to forward all of their calls to another extension or line.
You can enable the Call Forward On Busy feature in two ways:

By allowing the user to configure the call forward activation and its destination via the handset
(Steps 3-5).

By manually enabling the service (Steps 6-7).
 To set the Call Forward Unconditional feature:
1.
Select to which endpoint you want to apply the changes in the Select Endpoint drop-down menu at
the top of the window.
You have the choice between Default and all FXS endpoints your Mediatrix unit has.
2.
In the Unconditional section, define if you want to override the Call Forward Unconditional
parameters set in the Default configuration in the Endpoint Specific drop-down menu.
This menu is available only in the specific endpoints configuration.
Figure 179: Telephony – Call Forward Unconditional Section
2
3
4
5
6
7
3.
Enable the Call forward configuration via handset service by setting the Allow Activation via
Handset drop-down menu to Enable.
You also need to configure the activation and deactivation DTMF maps (steps 4 and 5).
If you select Disable, this does not disable the call forward, but prevents the user from activating or
deactivating the call forward service. The user will not be able to use the digits used to activate and
deactivate the call forward service.
4.
Define the digits that users must dial to start the service in the DTMF Map Activation field.
This field is available only in the Default configuration.
For instance, you could decide to put “*76” as the sequence to activate the service. This sequence
must be unique and follow the syntax for DTMF maps (see “Chapter 37 - DTMF Maps Configuration”
on page 369). Dialing this DTMF map does not have any effect unless the service’s status is
“enabled”.
The activating sequence is set for all the endpoints of the Mediatrix unit. You cannot have a different
sequence for each endpoint.
5.
Define the digits that users must dial to stop the service in the DTMF Map Deactivation field.
This field is available only in the Default configuration.
For instance, you could decide to put “*77” as the sequence to deactivate the service. This
sequence must be unique and follow the syntax for DTMF maps (see “Chapter 37 - DTMF Maps
Configuration” on page 369). Dialing this DTMF map does not have any effect unless the service’s
status is “enabled”.
The deactivating sequence is set for all the endpoints of the Mediatrix unit. You cannot have a
different sequence for each endpoint.
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6.
Set the status of the service in the Activation field to Inactive or Active.
Table 348: Activation State
State
Description
Inactive
The call forward service is not available on the telephone connected to the specific
endpoint. A call to this endpoint is not forwarded if the endpoint is busy.
Active
The call forward service is available on the telephone connected to the specific
endpoint. A call to the endpoint is forwarded to the specified destination if the
endpoint is busy. You must define the call forward destination in the Forwarding
Address field (Step 7). The call forward service behaves as if it is inactive if the
Forwarding Address is empty.
To let the user activate or deactivate this service with his or her handset, see steps 3, 4, and 5. In
that case, the field is automatically updated to reflect the activation status.
7.
Define the address to which forward incoming calls in the Forwarding Address field.
Accepted formats are:
•
telephone numbers (5551111)
•
SIP URLs such as ”scheme:user@host”. For instance, “sip:user@foo.com”.
This string is used literally, so cosmetic symbols (such as the dash in “555-xxxx”) should not be
present.
8.
Click Submit if you do not need to set other parameters.
Configuring Call Forward on Unconditional via Handset
When forwarding calls outside the system, a brief ring is heard on the telephone to remind the user that the
call forward service is active. The user can still make calls from the telephone.
 To forward calls:
1.
Take the receiver off-hook.
2.
Wait for the dial tone.
3.
Dial the sequence implemented to activate the call forward unconditional service.
This sequence could be something like *76.
4.
Wait for the stutter dial tone (three “beeps”) followed by the dial tone.
5.
Dial the number to which you want to forward your calls. Dial any access code if required.
6.
Wait for three “beeps” followed by a silent pause.
The call forward is established.
7.
Hang up your telephone.
 To check if the call forward has been properly established:
1.
Take the receiver off-hook.
2.
Wait for the dial tone.
3.
Dial your extension or telephone number.
The call is forwarded to the desired telephone number.
4.
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Hang up your telephone.
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Call Forward Unconditional
 To cancel the call forward:
1.
Take the receiver off-hook.
2.
Wait for the dial tone.
3.
Dial the sequence implemented to deactivate the call forward – unconditional service.
This sequence could be something like *77.
4.
Wait for the stutter dial tone (three “beeps”) followed by the dial tone.
The call forward is cancelled.
5.
384
Hang up your telephone.
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General Configuration
C
Software Configuration Guide
H A P T E R
39
Telephony Services
Configuration
This chapter describes how to set the following subscriber services:











Hook Flash Processing
Automatic call
Call completion
Delayed Hotline
Call Transfer
Call Waiting
Conference
Direct IP address call
Hold
Second call
Message Waiting Indicator
Some of the subscriber services are not supported on all Mediatrix unit models, so your specific model may
not have all subscriber services listed in this chapter.
You can use two types of configuration:


Default configurations that apply to all the endpoints of the Mediatrix unit.
Specific configurations that override the default configurations. You can define specific
configurations for each endpoint in your Mediatrix unit.
General Configuration
The General Configuration sub-section of the Services Configuration section allows you to define the Hook
Flash Processing feature.
Note: Performing a flash hook and pressing the flash button means the same thing. However, not all
telephone models have a flash button.
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General Configuration
 To set general services parameters:
1.
In the web interface, click the Telephony link, then the Services sub-link.
Figure 180: Telephony – Services Web Page
2
3
4
2.
Select to which endpoint you want to apply the changes in the Select Endpoint drop-down menu at
the top of the window.
You have the choice between Default and the interfaces of your Mediatrix unit. The number of
interfaces available vary depending on the Mediatrix unit model you have.
3.
In the General Configuration sub-section, define whether or not you want to override the general
services parameters set in the Default configuration in the Endpoint Specific drop-down menu.
This menu is available only in the specific endpoints configuration.
4.
Select how to process hook-flash detection in the Hook Flash Processing drop-down menu.
Hook flash processing allows hook flash signals to be transported over the IP network allowing to
use advanced telephony services. Users normally press the “flash” button of the telephone during
a call in progress to put this call on hold, transfer it, or even initiate a conference call.
You can define whether these subscriber services are handled by the unit or delegated to a remote
party. If services are to be handled by a remote party, a SIP INFO message is sent to transmit the
user's intention.
Note: The hook-flash processing attribute is not negotiated in SDP.
Table 349: Hook Flash Settings
Setting
Definition
Process Locally
The hook-flash is processed locally. The actual behaviour of the “flash”
button depends on which endpoint services are enabled for this
endpoint.
Transmit Using
Signaling Protocol
The hook-flash is processed by a remote party. The hook-flash event is
carried by a signaling protocol message. The actual behaviour of the
“flash” button depends on the remote party.
The hook-flash event is relayed as a SIP INFO message as described
in RFC 2976.
5.
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Click Submit if you do not need to set other parameters.
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Automatic Call
The automatic call feature allows you to define a telephone number that is automatically dialed when taking
the handset off hook.
When this service is enabled, the second line service is disabled but the call waiting feature is still functional.
The user can still accept incoming calls.
 To set the automatic call feature:
1.
Select to which endpoint you want to apply the changes in the Select Endpoint drop-down menu at
the top of the window.
You have the choice between Default and the interfaces of your Mediatrix unit. The number of
interfaces available vary depending on the Mediatrix unit model you have.
2.
In the Automatic Call sub-section, define whether or not you want to override the automatic call
parameters set in the Default configuration in the Endpoint Specific drop-down menu.
This menu is available only in the specific endpoints configuration.
Figure 181: Telephony – Automatic Call Section
2
3
4
3.
Enable the service by setting the Automatic Call Activation drop-down menu to Enable.
4.
Define the string to dial when the handset is taken off hook in the Automatic Call Target field.
Accepted formats are:
•
telephone numbers (5551111)
•
SIP URLs such as ”scheme:user@host”. For instance, “sip:user@foo.com”.
This string is used literally, so cosmetic symbols (such as the dash in “555-xxxx”) should not be
present.
5.
Click Submit if you do not need to set other parameters.
Call Completion
Note: This section applies only to the following models:
• Mediatrix 3208 / 3216
• Mediatrix 3308 / 3316
• Mediatrix 3716
• Mediatrix 3731
• Mediatrix 3732
• Mediatrix 3741
• Mediatrix 3742
• Mediatrix 4100 Series
• Mediatrix LP Series
• Mediatrix C7 Series
The call completion service allows you to configure the Completion of Calls on No Reply (CCNR) and
Completion of Calls to Busy Subscriber (CCBS) features.
CCBS allows a caller to establish a call with a “busy” callee as soon as this callee is available to take the call.
It is implemented by monitoring the activity of a UA and look for the busy-to-idle state transition pattern.
CCNR allows a caller to establish a call with an “idle” callee right after this callee uses his phone. It is
implemented by monitoring the activity of a UA and look for the idle-busy-idle state transition pattern.
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The information about the call completion is not kept after a restart of the EpServ service. This includes the
call completion activation in the Pots service and the call completion monitoring in the SipEp service.
 To set the call completion feature:
1.
Select to which endpoint you want to apply the changes in the Select Endpoint drop-down menu at
the top of the window.
You have the choice between Default and the interfaces of your Mediatrix unit. The number of
interfaces available vary depending on the Mediatrix unit model you have.
2.
In the Call Completion sub-section, define whether or not you want to override the call completion
parameters set in the Default configuration in the Endpoint Specific drop-down menu.
This menu is available only in the specific endpoints configuration.
Figure 182: Telephony – Call Completion Section
3
4
5
7
6
8
9
10
11
12
13
3.
Enable or disable the (CCBS) service by selecting the proper value in the Allow CCBS Activation
Via Handset drop-down menu.
You also need to configure the activation and deactivation DTMF maps (steps 4 and 7).
4.
If the CCBS service is enabled, define the digits that users must dial to start the service in the CCBS
DTMF Map Activation field.
This field is available only in the Default configuration.
You can use the same code in the CCNR DTMF Map Activation field.
For instance, you could decide to put “*92” as the sequence to activate the service. This sequence
must be unique and follow the syntax for DTMF maps (see “Chapter 37 - DTMF Maps Configuration”
on page 369). Dialing this DTMF map does not have any effect unless the service’s status is
“enabled”.
The activating sequence is set for all the endpoints of the Mediatrix unit. You cannot have a different
sequence for each endpoint.
5.
Enable or disable the (CCNR) service by selecting the proper value in the Allow CCNR Activation
Via Handset drop-down menu.
You also need to configure the activation and deactivation DTMF maps (steps 6 and 7).
6.
If the CCNR service is enabled, define the digits that users must dial to start the service in the CCNR
DTMF Map Activation field.
This field is available only in the Default configuration.
You can use the same code in the CCBS DTMF Map Activation field.
For instance, you could decide to put “*93” as the sequence to activate the service. This sequence
must be unique and follow the syntax for DTMF maps (see “Chapter 37 - DTMF Maps Configuration”
on page 369). Dialing this DTMF map does not have any effect unless the service’s status is
“enabled”.
The activating sequence is set for all the endpoints of the Mediatrix unit. You cannot have a different
sequence for each endpoint.
7.
Define the digits that users must dial to stop the CCBS and CCNR services in the DTMF Map
Deactivation field.
This field is available only in the Default configuration.
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For instance, you could decide to put “*94” as the sequence to deactivate the services. This
sequence must be unique and follow the syntax for DTMF maps (see “Chapter 37 - DTMF Maps
Configuration” on page 369). Dialing this DTMF map does not have any effect unless the service’s
status is “enabled”.
The deactivating sequence is set for all the endpoints of the Mediatrix unit. You cannot have a
different sequence for each endpoint.
8.
Define the delay, in minutes, after the call completion activation to automatically deactivate the call
completion if the call is not completed in the Expiration Timeout field.
This field is available only in the Default configuration.
9.
Select the call completion method to detect that the call completion destination is ready to complete
the call in the Method drop-down menu.
Table 350: Call Completion Method Parameters
Method
Monitoring Only
Desciption
The call completion only uses the monitoring method to detect that the
destination is ready to complete the call.
Monitoring And Polling The call completion only uses the monitoring method to detect that the
destination is ready to complete the call. The polling mechanism is used
if the call completion destination cannot be monitored.
This field is available only in the Default configuration.
The monitoring method consists of using the protocol signalling to detect the destination state
without using the call. When the destination is ready to complete the call, the local user is notified
that the call is ready to be completed and the call to the destination is initiated when the user is ready
to initiate the call.
The polling method consists of using periodic calls to the call completion destination until the
destination responds with a ringing or connect. Upon receiving these responses, the local user is
notified that the call is ready to be completed.
The polling mechanism can only be used for call completion to busy subscriber (CCBS).
The retransmission of the polling mechanism is configurable with
DefaultCallCompletionPollingInterval.
10.
Enable or disable the call completion auto reactivation in the Auto Reactivate drop-down menu.
This field is available only in the Default configuration.
When enabled, the call completion busy subscriber is automatically activated if the call initiated by
a call completion busy subscriber or call completion no response fails because of a busy
destination.
11.
Define the minimal delay to wait, in seconds, before executing a call completion after its activation
in the Auto Reactivate Delay field.
This field is available only in the Default configuration.
This delay only applies to call completion activated via the call completion auto reactivation feature
(See Step 9).
Media5 recommends to set a delay when the method to monitor the target state is based on the
target calls instead of its ability to answer a call.
If the timeout is set to 0 and the target is off hook, the FXS endpoint always rings to notify that the
call completion is ready to be completed. However the call is always busy and thus reactivated
without the possibility for the user to cancel the call completion. The call completion will continue
until the ringing or call completion timeout or if the target became ready to receive call.
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12.
General Configuration
Define how the call completion service needs to interpret the reception of a progress message with
early media in the Early Media Behaviour drop-down menu.
Table 351: Call Completion Early Media Behaviour Parameters
Parameter
Description
None
The progress message with early media is not considered as a busy or a ringing
response.
CCBS
The progress message with early media is interpreted as a busy response and the
CCBS can be activated on the call.
CCNR
The progress message with early media is interpreted as a ringing response and
the CCNR can be activated on the call.
This field is available only in the Default configuration.
13.
Define the delay, in seconds, between the calls to the call completion target used for the polling
mechanism in the Polling Interval field.
This field is available only in the Default configuration.
This parameter is used only if the Default Call Completion Method drop-down menu is set to
Monitoring And Polling.
14.
Click Submit if you do not need to set other parameters.
Special SIP Configuration
If you are using an Asterisk® IP PBX, it returns the error code 503 instead of 486 for a busy destination when
the call limit is reached. The following error mapping can be required:
1.
Go to the page SIP > Misc.
2.
Insert a new mapping (with the plus button) in the SIP To Cause Error Mapping section.
3.
Set the SIP code to 503 “Service Unavailable” and the cause to 17 “User busy”.
4.
Click Submit.
Using the Call Completion Services
The following are the various procedures to use these services on the user’s telephone.
 To start the CCBS (procedure 1)
The call has reached a busy destination and the busy tone is played.
1.
Dial the sequence implemented to enable the CCBS.
This sequence could be something like *92.
The confirmation tone is played.
2.
Hang up the telephone.
Alternatively, you can use procedure 2.
 To start the CCBS (procedure 2)
The call has reached a busy destination and the busy tone is played.
1.
Hang up the telephone.
2.
Take the receiver off-hook.
The dial tone is played
3.
Dial the sequence implemented to enable the CCBS.
This sequence could be something like *92.
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The confirmation tone is played.
4.
Hang up the telephone.
Alternatively, you can use procedure 1.
 To start the CCNR
The call has reached a destination but the call is still not yet established. A ring back or welcome message is
generally played at this moment.
1.
Hang up the telephone.
2.
Take the receiver off-hook.
The dial tone is played
3.
Dial the sequence implemented to enable the CCNR.
This sequence could be something like *93.
The confirmation tone is played.
4.
Hang up the telephone.
 To stop the CCBS or CCNR
1.
Take the receiver off-hook.
The dial tone is played
2.
Dial the sequence implemented to disable the CCBS and CCNR.
This sequence could be something like *93.
The confirmation tone is played.
3.
Hang up the telephone.
Note: The CCBS and CCNR cannot be started to complete a second call.
 When the call completion target is ready to receive a call:
1.
The telephone rings with the distinctive ringing “Bellcore-dr2” (0.8 On – 0.4 Off, 0.8 On – 4.0 Off).
2.
Hang up the telephone.
The call is initiated to the call completion destination.
Call Transfer
Note: This section applies only to the following models:
• Mediatrix 3208 / 3216
• Mediatrix 3308 / 3316
• Mediatrix 3716
• Mediatrix 3731
• Mediatrix 3732
• Mediatrix 3741
• Mediatrix 3742
• Mediatrix 4100 Series
• Mediatrix LP Series
• Mediatrix C7 Series
The Call Transfer service offers two ways to transfer calls:

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Blind Transfer
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
General Configuration
Attended Transfer
 To enable the Call Transfer services:
1.
Select to which endpoint you want to apply the changes in the Select Endpoint drop-down menu at
the top of the window.
You have the choice between Default and all FXS endpoints your Mediatrix unit has.
2.
In the Call Transfer sub-section, define whether or not you want to override the call transfer
parameters set in the Default configuration in the Endpoint Specific drop-down menu.
This menu is available only in the specific endpoints configuration.
Figure 183: Telephony – Call Transfer Web Page
3.
Enable the Blind Transfer service by setting the Blind Transfer Activation drop-down menu to
Enable.
The blind call transfer service is sometimes called Transfer without Consultation or Unattended
Transfer. It allows a user to transfer a call on hold to a still ringing (unanswered) call. The individual
at the other extension or telephone number does not need to answer to complete the transfer.
The call hold and second call services must be enabled for this service to work. See “Call Hold” on
page 403 and “Second Call” on page 404.
4.
Enable the Attended Transfer service by setting the Attended Transfer Activation drop-down menu
to Enable.
The attended call transfer service is sometimes called Transfer with Consultation. It allows a user
to transfer a call on hold to an active call. The individual at the other extension or telephone number
must answer to complete the transfer.
The call hold and second call services must be enabled for this service to work. See “Call Hold” on
page 403 and “Second Call” on page 404.
5.
Click Submit if you do not need to set other parameters.
Using Blind Call Transfer
The following is the procedure to use this service on the user’s telephone.
To configure the SIP Blind Transfer Method, see “SIP Blind Transfer Method” on page 314.
 To transfer a current call blind:
1.
Perform a Flash-Hook by pressing the “Flash” button on your analog telephone.
This puts the call on hold.
2.
Wait for the transfer tone (three “beeps”).
3.
Dial the number to which you want to transfer the call.
4.
Wait for the ringback tone, then hang up your telephone.
2
3
4
The call is transferred.
Once the transfer is executed, the remaining calls (call on hold and ringing call with third party) are
then connected together. The call on hold is automatically unheld and hears the ringback tone
provided by the third party's ringing.
You can also wait for the third party to answer if you want. In this case, the call transfer becomes
attended.
If you want to get back to the first call (the call on hold), you must perform a Flash-Hook.
You are back with the first call and the third party is released.
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Using Attended Call Transfer
The following is the procedure to use this service on the user’s telephone.
 To transfer a current call attended:
1.
Perform a Flash-Hook by pressing the “Flash” button on your analog telephone.
This puts the call on hold.
2.
Wait for the transfer tone (three “beeps”).
3.
Dial the number to which you want to transfer the call.
The third party answers.
4.
Hang up your telephone.
The call is transferred.
5.
If you want to get back to the first call (the call on hold), you must perform a Flash-Hook before the
target answers.
You are back with the first call and the third party is released.
Note: If the number to which you want to transfer the call is busy or does not answer, perform a Flash-Hook.
The busy tone or ring tone is cancelled and you are back with the first call.
Note: Attended call transfers can only be used to transfer a call already established. You cannot use the
Attended Call Transfer for an incoming call. For example, in this case where C is the incoming call.
1.
A calls B.
2.
B answers the call
3.
C calls B
4.
B puts A on hold (Flash hook) and answers C
5.
B hangs up the phone.
When B hangs up at step 5, A and C will not be connected. C will rather be released. B will ring and when
B answers, B will be in communication with A.
Call Waiting
Note: This section applies only to the following models:
• Mediatrix 3208 / 3216
• Mediatrix 3308 / 3316
• Mediatrix 3716
• Mediatrix 3731
• Mediatrix 3732
• Mediatrix 3741
• Mediatrix 3742
• Mediatrix 4100 Series
• Mediatrix LP Series
• Mediatrix C7 Series
The call waiting tone indicates to an already active call that a new call is waiting on the second line.
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Your users can activate/deactivate the call waiting tone for their current call. This is especially useful when
transmitting faxes. The user that is about to send a fax can thus deactivate the call waiting tone to ensure that
the fax transmission will not be disrupted by an unwanted second call. When the fax transmission is completed
and the line is on-hook, the call waiting tone is automatically reactivated.
 To set the Call Waiting services:
1.
Select to which endpoint you want to apply the changes in the Select Endpoint drop-down menu at
the top of the window.
You have the choice between Default and all FXS endpoints your Mediatrix unit has.
2.
In the Call Waiting sub-section, define whether or not you want to override the call waiting
parameters set in the Default configuration in the Endpoint Specific drop-down menu.
This field is available only in the specific endpoints configuration.
Figure 184: Call Waiting Section
2
3
4
5
6
3.
From the Call Waiting Activation drop-down menu, select Enable.
This permanently activates the call waiting tone. When receiving new calls during an already active
call, a special tone is heard to indicate that a call is waiting on the second line. The user can then
answer that call by using the “flash” button. The user can switch between the two active calls by
using the “flash” button.
The call hold service must be enabled for this service to work. See “Call Hold” on page 403.
If the user is exclusively using faxes, select Disable to permanently disable the call waiting tone.
4.
Define the digits that users must dial to disable the Call Waiting tone in the Cancel DTMF Map field.
This field is available only in the Default configuration. This allows a user who has call waiting
enabled to disable that service on the next call only. If, for any reason, the user wishes to undo the
cancel, unhook and re-hook the telephone to reset the service.
For instance, you could decide to put “*76” as the sequence to disable the call waiting tone. This
sequence must be unique and follow the syntax for DTMF maps (see “Chapter 37 - DTMF Maps
Configuration” on page 369). Dialing this DTMF map does not have any effect unless the service’s
status is “enabled”.
The deactivating sequence is set for all the endpoints of the Mediatrix unit. You cannot have a
different sequence for each endpoint.
5.
In the Activation DTMF Map field, define the digits that users must dial to activate the Call Waiting
service. Note that dialing this DTMF map does not have any effect unless the call waiting service's
status is 'enabled'.
6.
In the Deactivation DTMF Map field, define the digits that users must dial to deactivate the Call
Waiting service. Note that dialing this DTMF map does not have any effect unless the call waiting
service's status is 'enabled'.
7.
Click Apply if you do not need to set other parameters.
Using Call Waiting
The call waiting feature alerts the user if he or she is already on the telephone and a second call happens. A
“beep” (the call waiting tone) is heard and repeated every ten seconds to indicate there is a second incoming
call.
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 To put the current call on hold:
1.
Perform a Flash-Hook by pressing the “Flash” button on your analog telephone.
This puts the call on hold and the second line is automatically connected to your line.
2.
Answer the call on the second line.
 To switch from one line to the other:
1.
Perform a Flash-Hook each time you want to switch between lines.
 To terminate the first call before answering the second call:
1.
Hang up the telephone.
2.
Wait for the telephone to ring.
3.
Answer the telephone.
The second call is on the line.
Removing the Call Waiting Tone
You can temporarily deactivate the call waiting tone indicating a call is waiting. This is especially useful when
transmitting faxes. If you are about to send a fax, you can thus deactivate the call waiting tone to ensure that
the fax transmission is not disrupted by an unwanted second call. When the fax transmission is completed and
the line is on-hook, the call waiting tone is automatically reactivated.
 To deactivate the call waiting tone:
1.
Take the receiver off-hook.
2.
Wait for the dial tone.
3.
Dial the sequence implemented to deactivate the call waiting tone.
This sequence could be something like *76.
4.
Wait for the transfer tone (three “beeps”) followed by the dial tone.
The call waiting tone is disabled.
IMS-3GPP Communication Waiting
Upon receipt of a SIP INVITE with multipart/mixed content where a valid IMS communication waiting indicator
is correctly specified such as in this example:
INVITE sip:...
[...]
Content-Type: multipart/mixed;boundary=boundary1
[...]
--boundary1
Content-Type: application/vnd.3gpp.cw+xml
Content-Disposition: render;handling=optional
<?xml version="1.0"?>
<ims-cw xmlns="urn:3gpp:ns:cw:1.0">
<communication-waiting-indication/>
</ims-cw>
--boundary1
Content-Type: application/sdp
[...]
--boundary1--
The 180 Ringing response to this may contain a special header :
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Alert-Info: <urn:alert:service:call-waiting>
that is appended if all of the following are true :
1.
The INVITE contained the <communication-waiting-indication/> 3GPP option.
2.
The destination endpoint supports call waiting.
3.
The call waiting feature is enabled for this endpoint.
4.
The endpoint is currently in an active state (not ringing, not on hold, not on hook).
There are no variables to control this behaviour, it is always activated.
This header could be used by the server to notify the 2nd caller that the destination is currently busy
in a call but was notified of this new incoming call.
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Conference
Note: This section applies only to the following models:
• Mediatrix 3208 / 3216
• Mediatrix 3308 / 3316
• Mediatrix 3716
• Mediatrix 3731
• Mediatrix 3732
• Mediatrix 3741
• Mediatrix 3742
• Mediatrix 4100 Series
• Mediatrix LP Series
• Mediatrix C7 Series
The Conference Call service allows a user to link two or more calls together to form a single conversation,
called a conference.


Only 3-way conferences are currently supported.
A participant of the conference can put the conference on hold and attempt other calls. This
participant may then rejoin the conference at a later time by unholding it. The participant who
initiated the conference cannot put it on hold.
You must enable the call hold, second call and attended call transfer services for this service to work. See
“Call Hold” on page 403, “Second Call” on page 404, and “Call Transfer” on page 391.
The following is a conference call flow example:
Figure 185: Conference Call Flow
User
Agent
#2
(B)
INVITE (G.729)
Trying/ Ringing/200 OK
ACK
User
Agent
#1
(A)
User
Agent
#3
(C)
Flash Hook
INVITE (HOLD)
Trying /200 OK
ACK
INVITE (G.729 )
Trying/Ringing/200 OK
ACK
Flash Hook
Trying200 OK
ACK
INVITE (UNHOLD-G.729 )
Trying/200 OK
ACK
3-way Conference Call Established
DSP Limitation
The Mediatrix 4108, 4116, 4124, C7, LP16 and LP24 models currently suffer from a limitation of their DSPs.
When using a codec other than G.711, enabling Secure RTP (SRTP) and/or using conferences has an impact
on the Mediatrix unit’s overall performance as SRTP and conferences require CPU power. That is the reason
why there is a limitation on the lines that can be used simultaneously, depending on the codecs enabled and
SRTP. This could mean that a user picking up a telephone on these models may not have a dial tone due to
lack of resources in order to not affect the quality of ongoing calls. See “Security” on page 201 for more details
on SRTP limitations.
The DSPs offer channels as resources to the Mediatrix unit. The Mediatrix unit is limited to two conferences
per DSP.
Please note that:
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

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One FXS line requires one channel.
Each conference requires one additional channel
The Mediatrix 4108/C7 has one DSP
The Mediatrix 4116/LP16 have two DSPs
The Mediatrix 4124/LP24 have three DSPs
A total of eight channels per DSP are available when using unsecure communication, to be used between the
FXS lines and up to two conferences.
A total of six channels per DSP are available when using SRTP, to be used between the FXS lines and up to
two conferences.
Enabling the Conference Call Feature
You must enable this service before your users can use it.
 To enable the Conference service:
1.
Select to which endpoint you want to apply the changes in the Select Endpoint drop-down menu at
the top of the window.
You have the choice between Default and all FXS endpoints your Mediatrix unit has.
2.
In the Conference sub-section, define whether or not you want to override the conference
parameters set in the Default configuration in the Endpoint Specific drop-down menu.
This menu is available only in the specific endpoints configuration.
Figure 186: Conference Section
2
3
3.
Enable the service by setting the Conference Activation drop-down menu to Enable.
4.
Click Submit if you do not need to set other parameters.
Using an External Server for the Conference
This section describes configuration that is available only in the MIB parameters of the Mediatrix unit. You can
configure these parameters as follows:



by using a MIB browser
by using the CLI
by creating a configuration script containing the configuration variables
The Mediatrix unit can use an external server to mix the media of the conference. This conference type
requires the configuration of an external server. Using this type of conference does not affect the number of
simultaneous calls supported. You can use this feature only if the Conference service is enabled (see
“Enabling the Conference Call Feature” on page 398 for more details).
You can use two types of configuration:


Default configurations that apply to all the endpoints of the Mediatrix unit.
Specific configurations that override the default configurations.
You can define specific configurations for each endpoint in your Mediatrix unit. For instance, you
could enable a codec for all the endpoints of the Mediatrix unit and use the specific configuration
parameters to disable this same codec on one specific endpoint.
Using one or more specific parameter usually requires that you enable an override variable and set the specific
configuration you want to apply.
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 To use a server-based conference:
1.
In the EpServMIB, specify how to manage the conference by setting the defaultConferenceType
variable to the proper value.
You can also use the following line in the CLI or a configuration script:
This configuration only applies to a conference initiated by one of the unit's endpoint.
EpServ.defaultConferenceType="Value"
where Value may be one of the following:
Table 352: Conference Type Parameters
Value
Parameter
Description
100
Local
The media of the conference is locally mixed by the unit. This
conference type does not require any special support of the call peer
or server. Using this type of conference can reduce the number of
simultaneous calls supported.
200
ConferenceSer The unit uses an external server to mix the media of the conference.
ver
This conference type requires the configuration of an external server
(See Step 3). Using this type of conference does not affect the number
of simultaneous calls supported.
In Local mode, the number of participants is limited to the unit's model capacity. In
ConferenceServer mode, the number of participants is limited by the server's capacity.
2.
If you want to set a different conference type for one or more endpoints, set the following variables:
•
epSpecificConferenceEnableConfig variable for the specific endpoint you want to
configure to enable.
•
epSpecificConferenceType variable for the specific endpoint you want to configure to
the proper value.
You can also use the following lines in the CLI or a configuration script:
EpServ.epSpecificConference.EnableConfig[Id="Specific_Endpoint"]="1"
EpServ.epSpecificConference.Type[Id="Specific_Endpoint"]="Type"
where:
3.
•
Specific_Endpoint is the number of the endpoint you want to configure.
•
Value is the type as defined in Step 1.
If you have set the Conference type to ConferenceServer, in the SipEpMIB, set the
defaultConferenceType variable with the URI used in the request-URI of the INVITE sent to the
conference server as defined in RFC 4579.
You can also use the following line in the CLI or a configuration script:
SipEp.DefaultStaticConferenceServerUri="URI"
4.
If you want to set a different URI for one or more endpoints, set the following variables:
•
GwSpecificConferenceEnableConfig variable for the specific endpoint you want to
configure to enable.
•
GwSpecificConferenceServerUri variable for the specific endpoint you want to
configure to the proper value.
You can also use the following lines in the CLI or a configuration script:
EpServ.GwSpecificConference.EnableConfig[Id="Specific_Endpoint"]="1"
EpServ.GwSpecificConference.ServerUri[Id="Specific_Endpoint"]="URIValue"
where:
•
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•
URIValue is the URI you want to use.
Managing a Conference Call
If you are on the telephone with one person and want to conference with a third one, you can do so. In the
following examples, let’s assume that:




“A” is the conference initiator.

“E” is a fifth person that “C” wants to add to the conference in conferenceServer conference
type.
“B” is the person called on the first line.
“C” is the person called on the second line.
“D” is a fourth person that “A” wants to add to the conference in conferenceServer conference
type.
 To initiate a three-way conference (“A” and “B” already connected):
1.
“A” performs a Flash-Hook.
This puts “B” on hold and the second line is automatically connected. “A” hears a dial tone.
2.
“A” dials “C’s” number.
“A” and “C” are now connected.
3.
“A” performs another Flash-Hook.
The call on hold (“B”) is reactivated. “A” is now conferencing with “B” and “C”.
 “B” (or “C”) hangs up during the conference:
1.
“B” (or “C”) hangs up during the conference.
The conference is terminated, but the call between “A” and “C” (or “B”) is not affected and they are
still connected.
 “A” (conference initiator) hangs up during the conference:
1.
“A” hangs up.
The conference is terminated, both call "C" and "B" are also terminated.
 “A” wants to add a fourth member to the conference:
This is available only in the conferenceServer conference type.
1.
“A” performs a Flash-Hook.
“A” hears a dial tone. The second line is automatically connected. “B” and “C” are still in conference.
2.
“A” dials “D’s” number.
“A” and “D” are now connected.
3.
“A” performs another Flash-Hook.
“A” is now conferencing with “B”, “C”, and “D”.
 "C" wants to add a fifth member to the conference:
This is available only in the conferenceServer conference type.
1.
"C" performs a Flash-Hook.
"C" hears a dial tone. The second line is automatically connected. "A ", "B " and "D " are still in
conference.
2.
"C" dials "E's" number.
"C" and "E" are now connected.
3.
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"E" is now conferencing with "A ", "B", "C", and "D".
Delayed Hot Line
Note: This section applies only to the following models:
• Mediatrix 3208 / 3216
• Mediatrix 3308 / 3316
• Mediatrix 3716
• Mediatrix 3731
• Mediatrix 3732
• Mediatrix 3741
• Mediatrix 3742
• Mediatrix 4100 Series
• Mediatrix LP Series
• Mediatrix C7 Series
The delayed hot line feature (also called warm line) is used to make an automatic call to a specified address
on the two following conditions:

When the user picks up the phone but does not dial any digit. The configured destination is
automatically called upon picking up the phone and after waiting for the configurable number
of seconds without dialling.

When the user starts dialing but does not complete a valid number before the timeout set in the
Delayed Hotline Condition drop-down menu expires.
The condition on which the delayed hotline is activated is configurable. This feature thus places an automatic
call whenever the Delayed Hotline Condition timeout expires. It could be used as an alternative to the
emergency number (for instance, the 911 number in North America).
 To configure the basic delayed hot line feature:
1.
Select to which endpoint you want to apply the changes in the Select Endpoint drop-down menu at
the top of the window.
You have the choice between Default and all FXS endpoints your Mediatrix unit has.
2.
In the Delayed Hotline sub-section, define whether or not you want to override the delayed hotline
parameters set in the Default configuration in the Endpoint Specific drop-down menu.
This menu is available only in the specific endpoints configuration.
Figure 187: Delayed Hotline Section
2
3
3.
Enable the service by setting the Delayed Hotline Activation drop-down menu to Enable.
When the feature is disabled, a user picking up the phone but not pressing any telephone keys
hears the Receiver Off-Hook tone after the amount of time specified in the
digitMapTimeoutFirstDigit variable.
4.
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 To configure the delayed hotline activation condition:
1.
In the Delayed Hotline sub-section, select the condition(s) that activate the delayed hotline in the
Delayed Hotline Condition drop-down menu.
Figure 188: Delayed Hotline Section
1
Table 353: Delayed Hotline Conditions
Parameter
Description
FirstDtmfTimeout
The delayed hotline is activated when the timeout configured
in the First DTMF Timeout field of the Telephony > DTMF
Maps page elapses (“General DTMF Maps Parameters” on
page 372).
InterDtmfOrCompletionTimeout The delayed hotline is activated when the timeout configured
in the Completion Timeout field of the Telephony > DTMF
Maps page elapses or when the DTMFs collection fails
because the Inter DTMF Timeout parameter elapses
(“General DTMF Maps Parameters” on page 372).
AnyTimeout
2.
The delayed hotline is activated when the timeout configured
in the Completion Timeout field of the Telephony > DTMF
Maps page elapses and when the DTMFs collection fails
because the Inter DTMF Timeout parameter elapses
(“General DTMF Maps Parameters” on page 372).
Click Submit if you do not need to set other parameters.
 To configure the delayed hotline target:
1.
In the Delayed Hotline sub-section, set the destination (address or telephone number) that is
automatically called in the Delayed Hotline field.
Figure 189: Delayed Hotline Section
1
Accepted formats are:
•
telephone numbers (5551111)
•
SIP URLs such as ”scheme:user@host”. For instance, “sip:user@foo.com”.
This string is used literally, so cosmetic symbols (such as the dash in “555-xxxx”) should not be
present.
2.
Click Submit if you do not need to set other parameters.
Direct IP Address Call
The IP address call service allows a user to dial an IP address without the help of a SIP server. Using this
method bypasses any server configuration of your unit.
The user can dial an IP address and enter an optional telephone number. Note that the optional telephone
number is matched by using the same digit maps as a normal call.
The IP address call method can be used when a SCN user wants to reach a LAN endpoint.
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 To set the direct IP call feature:
1.
Select to which endpoint you want to apply the changes in the Select Endpoint drop-down menu at
the top of the window.
This menu is available only in the default endpoints configuration.
2.
Enable the service by setting the Direct IP Address Call drop-down menu to Enable.
Figure 190: Telephony – Direct IP Address Call Section
2
Dialing an IP Address
 To make an IP address call:
1.
Dial “**” (IP address prefix).
2.
Dial the numerical digits of the IP address and use the “*” for the “.” of the IP address.
3.
Dial “*” to terminate the IP address if you do not need to specify a phone number.
For instance, let’s say you want to reach a one-line access device or another LAN endpoint such as
an IP Phone with the IP address 192.168.0.23. You must then dial the following digits:
**192*168*0*23*
4.
If you need to specify the phone number of a specific line, dial “#” to terminate the IP address.
5.
Dial the telephone number of the specific line you want to reach.
For example, let’s say you want to reach the telephone connected to Line 2 of the Mediatrix unit with
the IP address 192.168.0.23. The phone number assigned to Line 2 of this Mediatrix unit is 1234.
You must then dial the following digits:
**192*168*0*23#1234
In this case, the Mediatrix unit sends an INVITE 1234@192.168.0.23.
Call Hold
Note: This section applies only to the following models:
• Mediatrix 3208 / 3216
• Mediatrix 3308 / 3316
• Mediatrix 3716
• Mediatrix 3731
• Mediatrix 3732
• Mediatrix 3741
• Mediatrix 3742
• Mediatrix 4100 Series
• Mediatrix LP Series
• Mediatrix C7 Series
The Call Hold service allows the user to temporarily put an existing call on hold, usually by using the “flash”
button of the telephone. The user can resume the call in the same way.
You must enable this service for the following services to work properly:





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 To enable the Call Hold service:
1.
Select to which endpoint you want to apply the changes in the Select Endpoint drop-down menu at
the top of the window.
You have the choice between Default and all FXS endpoints your Mediatrix unit has.
2.
In the Hold sub-section, define whether or not you want to override the call hold parameters set in
the Default configuration in the Endpoint Specific drop-down menu.
This menu is available only in the specific endpoints configuration.
Figure 191: Hold Section
2
3
3.
Enable the service by setting the Hold Activation drop-down menu to Enable.
4.
Click Submit if you do not need to set other parameters.
Using Call Hold
The following is the procedure to use this service on the user’s telephone.
 To put the current call on hold:
1.
Perform a Flash-Hook by pressing the “Flash” button on your analog telephone.
This puts the call on hold. You can resume the call in the same way.
Second Call
Note: This section applies only to the following models:
• Mediatrix 3208 / 3216
• Mediatrix 3308 / 3316
• Mediatrix 3716
• Mediatrix 3731
• Mediatrix 3732
• Mediatrix 3741
• Mediatrix 3742
• Mediatrix 4100 Series
• Mediatrix LP Series
• Mediatrix C7 Series
The Second Call service allows a user with an active call to put the call on hold, and then initiate a new call on
a second line. This service is most useful with the transfer and conference services.
The call hold service must be enabled for this service to work. See “Call Hold” on page 403.
You must enable this service for the following services to work properly:



Blind Transfer
Attended Transfer
Conference
 To enable the Second Call service:
1.
Select to which endpoint you want to apply the changes in the Select Endpoint drop-down menu at
the top of the window.
You have the choice between Default and all FXS endpoints your Mediatrix unit has.
2.
404
In the Second Call sub-section, define whether or not you want to override the second call
parameters set in the Default configuration in the Endpoint Specific drop-down menu.
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This menu is available only in the specific endpoints configuration.
Figure 192: Second Call Section
2
3
3.
Enable the service by setting the Second Call Activation drop-down menu to Enable.
4.
Click Submit if you do not need to set other parameters.
Using Second Call
The following is the procedure to use this service on the user’s telephone.
 To use the second call service:
1.
Perform a Flash-Hook by pressing the “Flash” button on your analog telephone.
This puts the call on hold and the second line is automatically connected to your line.
2.
Initiate the second call.
Message Waiting Indicator
Note: This section applies only to the following models:
• Mediatrix 3208 / 3216
• Mediatrix 3308 / 3316
• Mediatrix 3716
• Mediatrix 3731
• Mediatrix 3732
• Mediatrix 3741
• Mediatrix 3742
• Mediatrix 4100 Series
• Mediatrix LP Series
• Mediatrix C7 Series
This section describes configuration that is available only in the MIB parameters of the Mediatrix unit. You can
configure these parameters as follows:



by using a MIB browser
by using the CLI
by creating a configuration script containing the configuration variables
The Message Waiting Indicator (MWI) service alerts the user when new messages have been recorded on a
voice mailbox. It is enabled by default.
After the message is recorded, the server sends a message (SIP NOTIFY request) to the Mediatrix unit listing
how many new and old messages are available. The Mediatrix unit alerts the user of the new message in two
different ways:


The telephone’s LED blinks (if present). A FSK signal is sent on the FXS line.
A message waiting stutter dial tone replaces the normal dial tone when the user picks up the
FXS line.
Note: The message waiting state does not affect the Second Call feature. When in an active call,
performing a flash-hook to get access to the second line plays the usual dial tone.
The Mediatrix unit supports to receive SIP MWI notifications via SIP NOTIFY requests as defined in RFC 3842
but with the following limitations/diversions:
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Message Waiting Indicator

In addition to the SIP event string "message-summary" (RFC 3842), the string "simplemessage-summary" is accepted. The significations of those strings are identical.

In addition to the SIP content type string "simple-message-summary" (RFC 3842), the string
"message-summary" is accepted. The significations of those strings are identical.

Support of message-summary is not advertised in the SIP REGISTER.
Note that received SIP NOTIFY with an event different than "message-summary" or "simple-messagesummary" is not interpreted as a valid MWI notification.
You can use two types of configuration:


Default configurations that apply to all the endpoints of the Mediatrix unit.
Specific configurations that override the default configurations.
You can define specific configurations for each endpoint in your Mediatrix unit. For instance, you
could enable a codec for all the endpoints of the Mediatrix unit and use the specific configuration
parameters to disable this same codec on one specific endpoint.
Using one or more specific parameter usually requires that you enable an override variable and set the specific
configuration you want to apply.
 To disable the Message Waiting Indicator service:
1.
In the potsMIB, set the fxsDefaultMessageWaitingIndicatorActivation variable to the proper
value.
You can also use the following line in the CLI or a configuration script:
pots.fxsDefaultMessageWaitingIndicatorActivation="100"
If you want to reactivate the feature, use the following:
pots.fxsDefaultMessageWaitingIndicatorActivation="Value"
where Value may be one of the following:
Table 354: Message Waiting Indicator Parameters
Value
2.
Parameter
Description
100
Disabled
The user is not alerted of messages awaiting attention.
200
Tone
When messages are awaiting attention, the user is alerted by a
message waiting tone when picking up the handset.
300
Visual
When messages are awaiting attention, the user is alerted by a Visual
Message Waiting Indicator such as a blinking LED on the phone.
400
ToneAndVisual When messages are awaiting attention, the user is alerted by a Visual
Message Waiting Indicator such as a blinking LED on the phone, and
a message waiting tone when picking up the handset.
If you want to set a different activation for one or more endpoints, set the following variables:
•
fxsSpecificMessageWaitingIndicatorEnableConfig variable for the specific
endpoint you want to configure to enable.
•
fxsSpecificMessageWaitingIndicatorActivation variable for the specific endpoint
you want to configure to the proper value.
You can also use the following lines in the CLI or a configuration script:
pots.fxsSpecificMessageWaitingIndicator.EnableConfig[Id="Specific_Endpoint"]="1"
pots.fxsSpecificMessageWaitingIndicator.Activation[Id="Specific_Endpoint"]="Valu
e"
where:
406
•
Specific_Endpoint is the number of the endpoint you want to configure.
•
Value is the activation as defined in Step 1.
Dgw v2.0 Application
Distinctive Call Waiting Tone
Software Configuration Guide
Visual Message Waiting Indicator Type
You can configure how the Visual Message Waiting Indicator is sent on FXS lines.
 To configure the visual message waiting indicator type:
1.
In the potsMIB, set the fxsDefaultVisualMessageWaitingIndicatorType variable to the proper
value.
You can also use the following line in the CLI or a configuration script:
pots.fxsDefaultVisualMessageWaitingIndicatorType="Value"
where Value may be one of the following:
Table 355: Visual Message Waiting Indicator Type Parameters
Value
Parameter
Description
100
Fsk
A FSK signal is sent to activate the VMWI on the phone.
200
FskAndVoltage Both FSK signal and high voltage signal are used to activate the VMWI
on the phone.
Note: This parameter applies only to the following models:
•
Mediatrix 4108, 4116 and 4124
•
Mediatrix LP Series
Distinctive Call Waiting Tone
The distinctive call waiting tone configuration allows the administrator to modify the pattern of the tone.
Two variables are used:

ToneId: Allows the identification of the distinctive call waiting tone. If the distinctive ring callproperty matches the ToneId, the distinctive tone will be used.

Pattern: Describes the tone pattern.
A tone pattern contains:
1.
Frequencies
Up to 4 frequencies (f1 to f4) each with a power level can be defined. At least one frequency/power
pair must be defined. Frequency range is from 10 to 4000 Hz and Power level range is from -99 to
3 dbm.
The syntax is: f1=<frequency>:<power>
2.
States
Up to 8 states (s1 to s8) can be defined, each with an action, a set of frequencies, a duration and a
next state. At least one state must be described if the tone-pattern is not empty.
•
The action can be 'on', 'off' or 'CID' (for call waiting tones).
•
The duration of the state is from 10 to 56000 ms.
•
The tone is continuous if no time is specified.
The syntax is: s1=<action>:<frequency>:...:<frequency>:<duration>:<end-of-loop-indicator>:<next state>
3.
Loops
A set of states can be enclosed in a loop.
Dgw v2.0 Application
•
The starting state of a loop is marked with a loop counter (l=), the range is from 2 to
128.
•
The ending state of a loop is marked with an end-of-loop indicator (l).
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Call Statistics
The syntax is: l=<loop count>,<state definition>,...,<state definition (with end-of-loopindicator)>,<state definition>...
Examples:
•
Germany dialtone (continuous): "f1=350:-17,f2=440:-17,s1=on:f1:f2"
•
North America Recall dialtone (3 quick tones followed by a continuous tone): "f1=350:17,f2=440:-17,l=3,s1=on:f1:f2:100:s2,s2=off:100:l:s1,s3=on:f1:f2"
•
Australia ring back tone (on 400ms, off 200 ms, on 400 ms and off 2000 ms and
replay): "f1=425:-17,f2=400:-5,f3=450:5,s1=on:f1:f2:f3:400:s2,s2=off:200:s3,s3=on:f1:f2:f3:400:s4,s4=off:2000:s1"
Only two frequencies can be used by the Call Waiting tone.
The parameters can be set :



by using a MIB browser
by using the CLI
creating a configuration script containing the configuration variables
 To set the distinctive call waiting tone:
1.
2.
In the Tellf MIB set :
•
Tellf.DistinctiveCallWaitingPattern variable in the CallWaitingToneGroup table
•
Tellf.DistinctiveCallWaitingRingId variable in the CallWaitingToneGroup table.
•
or
Use the CLI or a configuration script:
•
Tellf.DistinctveCallWaiting[Index=value].Pattern=value
•
Tellf.DistinctveCallWaiting[Index=value].ToneId=value
Index value can vary from 1 to 4.
Call Statistics
This section describes how to access data available only in the MIB parameters of the Mediatrix unit. You can
display these parameters as follows:


by using a MIB browser
by using the CLI
The following are the call statistics the Mediatrix unit keeps. Statistics are updated at the end of each call.
Table 356: Call Statistics
MIB Variable
Statistics Description
IncomingCallsReceived
Number of incoming IP calls received on the endpoint
since service start.
IncomingCallsAnswered
Number of incoming IP calls answered on the endpoint
since service start.
IncomingCallsConnected Number of incoming IP calls that successfully completed
call setup signaling on the endpoint since service start.
IncomingCallsFailed
408
Number of incoming IP calls that failed to complete call
setup signaling on the endpoint since service start.
Dgw v2.0 Application
Default Outbound Priority Call Routing
Software Configuration Guide
Table 356: Call Statistics
MIB Variable
Statistics Description
OutgoingCallsAttempted
Number of outgoing IP calls attempted for the endpoint
since service start.
OutgoingCallsAnswered
Number of outgoing IP calls answered by the called
party for the endpoint since service start.
OutgoingCallsConnected Number of outgoing IP calls that successfully completed
call setup signaling for the endpoint since service start.
OutgoingCallsFailed
Number of outgoing IP calls that failed to complete call
setup signaling for the endpoint since service start.
CallsDropped
Number of IP calls, on the endpoint since service start,
that were successfully connected (incoming or
outgoing), but dropped unexpectedly while in progress
without explicit user termination.
TotalCallTime
Cumulative duration of all IP calls on the endpoint since
service start, in seconds.
 To display call statistics:
1.
In the epServMIB, go to the CallStatistics table.
You can also use the following line in the CLI:
get epServ.callStatistics
 To reset call statistics values to zero:
1.
In the epServMIB, set callStatistics.Reset to Reset for the endpoint to reset.
You can also use the following line in the CLI:
set epServ.callStatistics.Reset=Reset
2.
In the epServMIB, set callStatistics[EplId=callStatisticsEpId].Reset to Reset to reset
only one specific endpoint.
where:
•
callStatisticsEpId is the string that identifies the combination of an endpoint and a
channel. The endpoint name is the same as the EpId used to refer to endpoints in other
tables. On endpoints with multiple channels, the channel number must be appended at
the end of the endpoint name, separated with a dash.
You can also use the following line in the CLI:
set epServ.callStatistics[EplId=callStatisticsEpId].Reset=Reset
Examples:
Slot3/E1T1-12 refers to endpoint Slot3/E1T1, channel 12.
Phone-Fax1 refers to FXS endpoint Phone-Fax1 on a 4102s.
Port06 refers to FXS endpoint Port06 on 4108/4116/4124.
No channel number is appended to FXS endpoint strings because FXS lines do not support multiple
channels.
Default Outbound Priority Call Routing
This section describes configuration that is available only in the MIB parameters of the Mediatrix unit. You can
configure these parameters as follows:

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by using a MIB browser
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Chapter 39 - Telephony Services Configuration


Default Outbound Priority Call Routing
by using the CLI
by creating a configuration script containing the configuration variables
You can define how to route priority calls including emergency calls.
 To set the default outbound priority call routing:
1.
In the sipEpMIB, set the defaultOutboundPriorityCallRouting variable to the proper value.
You can also use the following line in the CLI or a configuration script:
sipEp.defaultOutboundPriorityCallRouting="Value"
where Value may be one of the following:
Table 357: Default Outbound Priority Call Routing Parameters
Value
410
Parameter
Description
100
Normal
Sends the call using normal SIP call routing to the outbound proxy (if
defined) and to the target host (usually the SIP server).
200
SkipOutbound
Proxy
Sends the call directly to the configured server skipping the outbound
proxy.
Dgw v2.0 Application
Current Tone Definition
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Software Configuration Guide
H A P T E R
40
Tone Customization Parameters
Configuration
This chapter describes how to override the pattern for a specific tone defined for the selected country. (For
more details on Tone Definition, refer the Reference Guide at http://www.media5corp.com/documentation.
It covers the following topics:


Current Tone Definition
Tone Override
Current Tone Definition
The Tone Customization page allows you to both see the current definition and override the pattern of the
following tones:














Busy
Call Waiting
Confirmation
Congestion
Dial
Hold
Intercept
Message Waiting
Preemption
Reorder
Ringback
Receiver Off Hook (ROH)
Special Information Tone (SIT)
Stutter
This includes the number of frequencies used, the tone value in Hertz (Hz), its power in dBm, as well as the
states configured.
 To see the current definition of a tone:
1.
In the web interface, click the Telephony link, then the Tone Customization sub-link.
Figure 193: Telephony – Tone Customization Web Page
2
2.
Dgw v2.0 Application
Select the proper tone to see in the Select Tone drop-down menu at the top of the window.
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Tone Override
The Current Tone Definition and Current Tone States sections describe the current definition of the
selected tone.
Tone Override
You can override the pattern for a specific tone. This is done in two sections:
Table 358: Tone Override Sections
State
Description
Overridden Tone Definition Allows you to define up to four frequencies (F1 to F4). You must enter at least
one frequency.
Overridden Tone States
Description of the tone state. You can define up to eight states. You must
enter at least one state.
 To override the pattern of a tone:
1.
Select which tone you want to override in the Override Current Tone Values drop-down menu.
Figure 194: Tone Override Sections
1
2
5
2.
6
3
7
8
9
4
•
You can use the current values of the selected tone as a starting point for your
customization by clicking the Copy Current Tone Definition to Overridden button.
•
You can clear all override fields by clicking the Reset Overridden Values button.
In the Overridden Tone Definition section, define the value of the proper Frequency used in the
corresponding Value field.
The value is in Hz. The range is from 10 Hz to 4000 Hz.
Note: You can use only two frequencies for the Call Waiting tone.
3.
Define the power level of the proper Frequency in dBm in the corresponding Power field.
The range is from -99 dBm to 3 dBm.
4.
If applicable, enter a value for the loop counter in the Loop Count field.
The range is from 2 to 128. This value will be used in Step 8.
Note: You can use only one loop count for the Call Waiting tone.
5.
412
In the Overridden Tone States section, set the corresponding On/Off drop-down menu with the
proper value for each state.
Dgw v2.0 Application
Tone Override
Software Configuration Guide
•
On means the corresponding state plays a tone.
•
Off means the corresponding state does not play a tone.
•
CID means the moment where the Caller-ID will be sent to the analog port. This options
is available only for the Call Waiting tone.
You may also want to perform the following operations:
6.
•
To add a state, click the
button at the bottom of the Overridden Tone States section.
•
To remove a state, click the
button at the bottom of the Overridden Tone States
section. This removes the last state in the list.
For the On states, select the frequency to play in the corresponding Frequencies column.
The frequencies defined in the Overridden Tone Definition section are listed as clickable buttons.
You can use from one to four frequencies. A blue button indicates that the frequency is selected.
7.
Set the corresponding Duration field with the number of times, in ms, to perform the action of the
state.
The range is from 10 ms to 56000 ms. The tone stays indefinitely in the state (continuous) if no time
is specified.
8.
In the corresponding Loop drop-down menu, select whether or not to stop looping between states
after a number of loops defined in Step 4.
When the number of loops is reached, the next state is s(n+1) for the state s(n) instead of the state
defined in the Next State drop-down menu.
9.
In the corresponding Next State drop-down menu, select the next tone state to use when the time
has elapsed.
This value is not available if the Duration field is empty.
10.
Dgw v2.0 Application
Click Submit if you do not need to set other parameters.
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Tone Override
Dgw v2.0 Application
MP3 File Download Server
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H A P T E R
41
Music on Hold Parameters
Configuration
This chapter describes how to configure the Music on Hold (MoH) parameters.


MP3 file download server setup.
Music on Hold configuration.
MP3 File Download Server
To download a MP3 file, you may need to setup the following applications on your computer:


TFTP server with proper root path
HTTP server with proper root path
Configuring the TFTP Server
When you perform a MP3 file download by using the TFTP (Trivial File Transfer Protocol) protocol, you must
install a TFTP server running on the PC designated as the TFTP server host. It is assumed that you know how
to set the TFTP root path. If not, refer to your TFTP server’s documentation.
Configuring the HTTP Server
When you to perform a MP3 file download by using the HTTP protocol, you must install a HTTP server running
on the PC designated as the server host. It is assumed that you know how to set the root path. If not, refer to
your HTTP server’s documentation.
Music on Hold Configuration
The Music on Hold sub-page of the Telephony page allows you to configure the music (in the form of an MP3
file) that plays when a local user has been put on hold. Note that transfers exceeding 5 minutes are cancelled.
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Chapter 41 - Music on Hold Parameters Configuration
Music on Hold Configuration
 To set the Music on Hold parameters:
1.
In the web interface, click the Telephony link, then the Music on Hold sub-link.
Figure 195: Telephony – Music on Hold Web Page
2
3
4
5
2.
In the Music On Hold Configuration section, indicate whether or not the unit should play music when
being put on hold in the Streaming drop-down menu.
When enabled, music is played toward the telephony side when being put on hold from the network
side.
3.
In the Transfer Configuration section, enter the URL to the MP3 file to use in the URL field.
This file is loaded when the Mediatrix unit starts and reloaded every time the Reload Interval value
elapses (see Step 5). It must be smaller than 1024 Kilobytes unless otherwise specified in a
customer profile.
The MP3 file downloaded must be encoded with a sampling rate of 8000 Hz (only available through
MPEG version 2.5) and in mono channel mode. All other types of file will be rejected. The decoding
output will be in mono channel mode, with a sample rate of 8000 Hz and with 8 bits per sample.
You can use the following supported protocols to transfer the file:
•
HTTP: HyperText Transfer Protocol.
•
TFTP: Trivial File Transfer Protocol.
URLs using any other transfer protocol are invalid.
Note: The HTTP protocol does not support spaces between characters in the URL.
Examples of valid URLS:
•
http://www.myserver.com/myfile.mp3
•
tftp://myserver.com:69/myfolder/myfile.mp3
When the port is not included in the URL, the default port for the chosen protocol is used.
HTTP supports basic or digest authentication mode as described in RFC 2617.
If you have selected HTTP, please note that your server may activate some caching mechanism for
the MP3 download. This mechanism caches the initial MP3 download for later processing, thus
preventing changes of the original MP3.
4.
If your server requires authentication when downloading the MP3, set the following:
•
The user name in the User Name field.
•
The password in the Password field.
Caution: The User Name and Password fields are not accessible if you have the User or Observer access
right. See “Users” on page 537 for more details.
416
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Music on Hold Configuration
5.
Software Configuration Guide
Set the time, in hours, between attempts to load the MP3 file in the Reload Interval field.
If you enter the value 0, this means that the unit loads the file only once at unit startup. Any other
value between 1 and 6000 is the number of hours between automatic reloads of the file. When a
manual file download is triggered, the counter is not reset so the next reload will happen at the same
time.
6.
Dgw v2.0 Application
If you do not need to set other parameters, do one of the following:
•
To save your settings without transferring the MP3 file, click Submit.
•
To save your settings and transfer the MP3 file now, click Submit & Transfer Now.
•
To save your settings and stop a file transfer in progress, click Submit & Cancel
Transfer.
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Music on Hold Configuration
Dgw v2.0 Application
Country Configuration
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Software Configuration Guide
H A P T E R
42
Country Parameters
Configuration
This chapter describes how to


configure the country information:
•
Select a specific country.
•
Additional country settings.
set the input and output offset
Country Configuration
The Misc sub-page of the Telephony page allows you to configure the country in which the unit is located. It
also allows the user to change the Input and Output Offsett.
 To select a specific country:
1.
In the Web interface, go toTelephony / Misc
Figure 196: Telephony – Misc Web Page
2.
In the Country section, from the Country Selection drop-down menu, select the country in which the
Mediatrix unit is located
It is very important to set the country in which the unit is used because a number of parameter
values are set according to this choice, such as tones, rings, impedances, and line attenuations.
See Reference Guide for more information on these country-specific settings.
3.
Click Apply.
 To set the Input Offset
Dgw v2.0 Application
1.
In the Web interface, go toTelephony/Misc
2.
From the Select Endpoint selection list, select an endpoint.
3.
In the User Gain section, enter the Input Offset value in the Input Offset field.
4.
Click Apply.
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Chapter 42 - Country Parameters Configuration
Country Configuration
 To set the Output Offset
420
1.
In the Web interface, go to Telephony/Misc
2.
From the Select Endpoint selection list, select an endpoint.
3.
In the User Gain section, enter the Output Offset value in the Output Offset field.
4.
Click Apply.
Dgw v2.0 Application
Additional Country Settings
Software Configuration Guide
Additional Country Settings
This section describes configuration that is available only in the MIB parameters of the Mediatrix unit. You can
configure these parameters as follows:



by using a MIB browser
by using the CLI
by creating a configuration script containing the configuration variables
Default vs. Specific Configurations
You can use two types of configuration:


Default configurations that apply to all the endpoints of the Mediatrix unit.
Specific configurations that override the default configurations.
You can define specific configurations for each endpoint in your Mediatrix unit. For instance, you
could enable a codec for all the endpoints of the Mediatrix unit and use the specific configuration
parameters to disable this same codec on one specific endpoint.
Using one or more specific parameter usually requires that you enable an override variable and set the specific
configuration you want to apply.
Input/Output User Gain
The user gain allows you to modify the input and output sound level of the Mediatrix unit.
Caution: Use these settings with great care. Media5 recommends not to modify the user gain variables
unless absolutely necessary because default calibrations may no longer be valid.
Modifying user gains may cause problems with DTMF detection and voice quality – using a high user gain
may cause sound saturation (the sound is distorted). Furthermore, some fax or modem tones may no longer
be recognized. The user gains directly affect the fax communication quality and may even prevent a fax to
be sent.
You can compensate with the user gain if there is no available configuration for the country in which the
Mediatrix unit is located. Because the user gain is in dB, you can easily adjust the loss plan, e.g., if you need
an additional 1 dB for analog to digital, put 1 for user gain output.
You can use two types of configuration as described in “Default vs. Specific Configurations” on page 421.
 To set user gain variables:
1.
In the telIfMIB, locate the countryCustomizationUserGainGroup folder.
2.
On a call involving a SIP terminal and an FXS terminal, raising the output offset will raise the volume
perceived on the FXS terminal. Define the default user output gain offset in dB in the
defaultCountryCustomizationUserGainOutputOffset variable.
You can also use the following line in the CLI or a configuration script:
telIf.defaultCountryCustomizationUserGainOutputOffset="Value"
Values range from -12 dB to +12 dB. However, going above +6 dB may introduce clipping/distortion
depending on the country selected.
3.
If you want to set a different output gain offset for one or more interfaces, set the following variables:
•
specificCountryCustomizationUserGainEnableConfig variable for the specific
interface you want to configure to enable.
•
specificCountryCustomizationUserGainOutputOffset variable for the specific line
you want to configure.
You can also use the following lines in the CLI or a configuration script:
telIf.specificCountryCustomizationUserGain.EnableConfig[InterfaceId="Interface"]
="1"
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Chapter 42 - Country Parameters Configuration
Additional Country Settings
telIf.specificCountryCustomizationUserGain.OutputOffset[InterfaceId="Interface"]
="Value"
where:
4.
•
Interface is the name of the interface you want to configure (for instance, Slot2/Pri1).
•
Value is the output gain offset.
On a call involving a SIP terminal and an FXS terminal, raising the input offset will raise the volume
perceived on the SIP terminal. Define the default user input gain offset in dB in the
defaultCountryCustomizationUserGainInputOffset variable.
You can also use the following line in the CLI or a configuration script:
telIf.defaultCountryCustomizationUserGainInputOffset="Value"
Values range from -12 dB to +12 dB. However, going above +6 dB may introduce clipping/distortion
depending on the country selected.
5.
If you want to set a different input gain offset for one or more interfaces, set the following variables:
•
specificCountryCustomizationUserGainEnableConfig variable for the specific
interface you want to configure to enable.
•
specificCountryCustomizationUserGainInputOffset variable for the specific line
you want to configure.
You can also use the following lines in the CLI or a configuration script:
telIf.specificCountryCustomizationUserGain.EnableConfig[InterfaceId="Interface"]
="1"
telIf.specificCountryCustomizationUserGain.InputOffset[InterfaceId="Interface"]=
"Value"
where:
6.
•
Interface is the name of the interface you want to configure (for instance, Slot2/Pri1).
•
Value is the input gain offset.
Restart the TelIf service by accessing the scmMIB and setting the serviceCommandsRestart
variable for the TelIf service to restart.
You can also use the following line in the CLI or a configuration script:
scm.serviceCommands.Restart[Name=TelIf]="10"
Dialing Settings
Dialing settings allow you to configure how the Mediatrix unit dials numbers.
When selecting a country (see “Country Configuration” on page 419 for more details), each country has default
dialing settings. However, you can override these values and define your own dialing settings.
You can use two types of configuration as described in “Default vs. Specific Configurations” on page 421.
 To set the dialing settings:
1.
In the telIfMIB, locate the countryCustomizationDialingGroup folder.
2.
Set the defaultCountryCustomizationDialingOverride variable to enable.
You can also use the following line in the CLI or a configuration script:
telIf.specificCountryCustomizationDialing.EnableConfig[InterfaceId="Interface"]=
"1"
where Interface is the name of the interface you want to configure (for instance, Slot2/Pri1).
This allows overriding the default country settings.
3.
If you want to change the override status for one or more interfaces, set the following variables:
•
specificCountryCustomizationDialingEnableConfig variable for the specific
interface you want to configure to enable.
•
specificCountryCustomizationDialingOverride variable for the specific interface
you want to configure to enable.
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You can also use the following lines in the CLI or a configuration script:
telIf.specificCountryCustomizationDialing.EnableConfig[InterfaceId="Interface"]=
"1"
telIf.specificCountryCustomizationDialing.Override[InterfaceId="Interface"]="1"
where Interface is the name of the interface you want to configure (for instance, Slot2/Pri1).
4.
Set an inter-digit dial delay in the defaultCountryCustomizationDialingInterDtmfDialDelay
variable.
You can also use the following line in the CLI or a configuration script:
telIf.defaultCountryCustomizationDialing.InterDtmfDialDelay="Value"
This is the delay, in milliseconds (ms), between two DTMFs when dialing the destination phone
number. Values range from 50 ms to 600 ms.
5.
If you want to set a different inter-digit dial delay for one or more interfaces, set the following
variables:
•
specificCountryCustomizationDialingEnableConfig variable for the specific
interface you want to configure to enable.
•
specificCountryCustomizationDialingInterDtmfDialDelay variable for the
specific interface you want to configure.
You can also use the following lines in the CLI or a configuration script:
telIf.specificCountryCustomizationDialing.EnableConfig[InterfaceId="Interface"]=
"1"
telIf.specificCountryCustomizationDialing.InterDtmfDialDelay[InterfaceId="Slot3/
Bri3"]="Value"
where Interface is the name of the interface you want to configure (for instance, Slot2/Pri1).
6.
Set the DTMF duration value in the defaultCountryCustomizationDialingDtmfDuration
variable.
You can also use the following line in the CLI or a configuration script:
telIf.defaultCountryCustomizationDialing.DtmfDuration="Value"
This is the duration, in milliseconds (ms), a DTMF is played when dialing the destination phone
number. Values range from 50 ms to 600 ms.
7.
If you want to set a different DTMF duration value for one or more interfaces, set the following
variables:
•
specificCountryCustomizationDialingEnableConfig variable for the specific
interface you want to configure to enable.
•
specificCountryCustomizationDialingDtmfDuration variable for the specific
interface you want to configure.
You can also use the following lines in the CLI or a configuration script:
telIf.specificCountryCustomizationDialing.EnableConfig[InterfaceId="Interface"]=
"1"
telIf.specificCountryCustomizationDialing.DtmfDuration[InterfaceId="Interface"]=
"Value"
8.
Set the delay, in milliseconds, between two MFR1s when dialing on the interface in the
DefaultCountryCustomizationDialingInterMfR1DialDelay variable.
See “Chapter 25 - E&M CAS Configuration” on page 223 for more details on MFR1 signalling.
You can also use the following line in the CLI or a configuration script:
9.
Set the delay, in milliseconds, between two MFR1s when dialing on the interface by putting the
following line in the configuration script:
telIf.defaultCountryCustomizationDialing.InterMfR1DialDelay="Value"
Values range from 50 ms to 600 ms.
10.
Dgw v2.0 Application
If you want to set a different delay value for one or more interfaces, set the following variables:
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Additional Country Settings
telIf.specificCountryCustomizationDialing.EnableConfig[InterfaceId="Interface"]=
"1"
telIf.specificCountryCustomizationDialing.InterMfR1DialDelay[InterfaceId="Interf
ace"]="Value"
11.
Set the duration, in milliseconds, of a MFR1 when dialling on the interface in the
DefaultCountryCustomizationDialingMfR1Duration variable.
See “Chapter 25 - E&M CAS Configuration” on page 223 for more details on MFR1 signalling.
You can also use the following line in the CLI or a configuration script:
12.
Set the duration, in milliseconds, of a MFR1 when dialing on the interface by putting the following
line in the configuration script:
telIf.DefaultCountryCustomizationDialing.MfR1Duration="Value"
Values range from 50 ms to 600 ms.
13.
If you want to set a different duration value for one or more interfaces, set the following variables:
telIf.specificCountryCustomizationDialing.EnableConfig[InterfaceId="Interface"]=
"1"
telIf.specificCountryCustomizationDialing.MfR1Duration[InterfaceId="Interface"]=
"Value"
14.
Restart the TelIf service by accessing the scmMIB and setting the serviceCommandsRestart
variable for the TelIf service to restart.
You can also use the following line in the CLI or a configuration script:
scm.serviceCommands.Restart[Name=TelIf]="10"
Fax Calling Tone Detection
You can enable the fax calling tone (CNG tone) detection.
You can use two types of configuration as described in “Default vs. Specific Configurations” on page 421.
 To enable fax calling tone detection:
1.
In the telIfMIB, locate the machineDetectionGroup folder.
2.
Set the defaultMachineDetectionCngToneDetection variable to enable.
You can also use the following line in the CLI or a configuration script:
telIf.defaultMachineDetection.CngToneDetection="1"
Upon recognition of the CNG tone, the Mediatrix unit switches the communication from voice mode
to fax mode and the CNG is transferred by using the preferred fax codec. This option allows for
quicker fax detection, but it also increases the risk of false detection.
If you do not want the Mediatrix unit to detect the fax calling tone, set the variable to disable(0). In
this case, the CNG tone does not trigger a transition from voice to data and the CNG is transferred
in the voice channel. With this option, faxes are detected later, but the risk of false detection is
reduced.
3.
If you want to set a different calling tone detection setting for one or more interfaces, set the
following variables:
•
specificMachineDetectionEnableConfig variable for the specific interface you want
to configure to enable.
•
specificMachineDetectionCngToneDetection variable for the specific interface you
want to configure.
You can also use the following lines in the CLI or a configuration script:
telIf.specificMachineDetection.EnableConfig[InterfaceId="Interface"]="1"
telIf.specificMachineDetection.CngToneDetection[InterfaceId="Interface"]="Value"
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CDR (Call Detail Record)
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H A P T E R
43
Call Detail Record
This chapter describes how to configure call detail record:
CDR (Call Detail Record)
Call detail record (CDR) in VoIP contains information about recent system usage such as the identities of
sources (points of origin), the identities of destinations (endpoints), the duration of each call, the total usage
time in the billing period and many others.
The Misc sub-page of the Telephony page allows you to configure the CDR parameters.
 To set the CDR parameters:
1.
In the Call Detail Record section of the Misc page, set the host name and port number of the device
that archives CDR log entries in the Syslog Remote Host field.
Specifying no port (or port 0) sends notifications to port 514.
Figure 197: CDR Call Detail Record Section
1
3
2.
2
Specify the format of the syslog Call Detail Record in the Syslog Format field.
The formal syntax description of the protocol is as follows:
Precision=DIGIT
Width=DIGIT
MacroId=(ALPHA / "_")
Macro=%[Width]|[.Precision]|[Width.Precision]MacroId
The Width field is the minimum width of the converted argument. If the converted argument has
fewer characters than the specified field width, then it is padded with spaces. If the converted
argument has more characters than the specified field width, the field width is extended to whatever
is required.
The Precision field specifies the maximum number of characters to be printed from a string.
Examples :
sipid=SipUser001
CDR Log: %sipid
CDR Log: %15sipid
CDR Log: %15.5sipid
CDR Log: %.5sipid
-->
-->
-->
-->
CDR
CDR
CDR
CDR
Log
Log
Log
Log
:
:
:
:
SipUser001
SipUser001
SipUs
SipUs
Call Detail Record predefined macros.
Control characters:
Table 359: Control Character
Character
Dgw v2.0 Application
Value
%%
%
\n
Split message
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CDR (Call Detail Record)
Call detail record macros:
Table 360: Call Detail Record Macros
Macro
Value
%id
CDR ID. The CDR ID is unique. The ID is incremented by one each time it is
represented in a CDR record
%sipid
SIP call ID. Blank if no SIP interface was used during the call.
%ocgnum Original calling number. Calling number as received by the unit.
%cgnum
Calling number. Calling number after manipulation by the call router.
%ocdnum Original called number. Called number as received by the unit.
426
%cdnum
Called number. Called number after manipulation by the call router.
%oiname
Original Interface name. Interface on which the call was received. Ex. isdn-Slot2/
Pri1.
%diname
Destination interface name. Interface on which the call was relayed. Ex. SIPDefault
%chan
Channel number. Blank if no PRI/BRI interface was used during the call. If 2 PRI/
BRI interface were involved, display the originating interface.
%sipla
SIP local IP address.
%sipra
SIP remote IP address or FQDN (next hop).
%siprp
SIP remote port (next hop).
%mra
Media remote IP address. Source IP address of incoming media stream. If the
stream was modified during the call, display the last stream.
%mrsp
Media remote port. Source port of incoming media stream. If the stream was
modified during the call, display the last stream.
%mdrp
Media remote port. Destination port of outgoing media stream. If the stream was
modified during the call, display the last stream.
%tz
Local time zone
%cd
Call duration (in seconds) (connect/disconnect).
%sd
Call duration (in seconds) (setup/connect).
%pdd
Post dial delay (in seconds) (setup/progress).
%css
Call setup second (local time)
%csm
Call setup minute (local time)
%csh
Call setup hour (local time)
%csd
Call setup day (local time)
%csmm
Call setup month (local time)
%csy
Call setup year (local time)
%ccs
Call connect second (local time)
%ccm
Call connect minute (local time)
%cch
Call connect hour (local time)
%ccd
Call connect day (local time)
%ccmm
Call connect month (local time)
%ccy
Call connect year (local time)
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CDR (Call Detail Record)
Software Configuration Guide
Table 360: Call Detail Record Macros (Continued)
Macro
3.
Value
%cds
Call disconnect second (local time)
%cdm
Call disconnect minute (local time)
%cdh
Call disconnect hour (local time)
%cdd
Call disconnect day (local time)
%cdmm
Call disconnect month (local time)
%cdy
Call disconnect year (local time)
%miptxc
IP Media last transmitted codec
%miptxp
IP Media last transmitted p-time
%dr
Disconnect reason (ISDN reason codes with ISUP SIP mapping)
%rxp
Received media packets. Excluding T.38.
%txp
Transmitted media packets. Excluding T.38.
%rxpl
Received media packets lost. Excluding T.38.
%rxmd
Received packets mean playout delay (ms, 2 decimals). Excluding T.38.
%rxaj
Received packets average jitter (ms, 2 decimals). Excluding T.38.
%sipdr
SIP disconnect or rejection reason.
Set the Syslog facility used by the unit to route the Call Detail Record messages in the Syslog
Facility field.
The application can use Local0 through Local7.
4.
Dgw v2.0 Application
Click Submit if you do not need to set other parameters.
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Call Router Parameters
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Introduction
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H A P T E R
44
Call Router Configuration
This chapter describes the call router service.








Introduction to the call router’s parts and types supported.
Routes parameters.
Mappings parameters.
Call signalling parameters.
SIP headers translation parameters.
Call properties translation parameters.
Hunt table parameters.
SIP Redirects parameters.
Introduction
The Mediatrix unit’s call router allows you to route calls between interfaces. Based on a set of routing criteria,
the call router determines the destination (interface) for every incoming call. The forwarding decisions are
based on the following tables:
Table 361: Call Router Table Types
Table
Description
Routing
The routing table contains one or more routes. Each route associates a destination to a call
that matches a set of criteria. See “Routes” on page 449 for more details.
Mapping
The mapping table contains one or more mapping types and expressions. A mapping
modifies call properties such as the calling and called party numbers according to the network
requirements. These mappings are specifically called within a route. See “Mappings” on
page 455 for more details.
Call
Signalling
Call signalling specifies how to set up a call to the destination Mediatrix unit or 3rd party
equipment. Call signalling properties are assigned to a route and used to modify the
behaviour of the call at the SIP protocol level. See “Signalling Properties” on page 465 for
more details.
SIP
A SIP headers translation overrides the default value of SIP headers in an outgoing SIP
Headers
message. See “SIP Headers Translations” on page 469 for more details.
Translation
Call
A call properties translation overrides the default value of call properties in an incoming SIP
Properties message. See “Call Properties Translations” on page 472 for more details.
Translation
Hunt
The hunt table contains one or more hunt entries, each with a set of possible destinations. A
hunt tries the destinations until one of the configured destinations accepts the call. See “Hunt
Service” on page 475 for more details.
SIP
Redirects
The SIP Redirects table allows configuring of SIP redirections that can be used as Route
destinations. When the Route source is a SIP interface, incoming SIP Invites are replied with
a 302 “Moved Temporarily” SIP response. See “SIP Redirects” on page 483 for more details.
When a new call comes from one of the Mediatrix unit interfaces, it is redirected to the routing table. The
following figure illustrates the Mediatrix unit call router:
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Chapter 44 - Call Router Configuration
Introduction
Figure 198: Call Routing
Interfaces
Call Router
Route 1
Signalling
Properties 1
Route 2
Route 3
Mapping 1
Hunt 1
Translation 1
(SIP Headers or
call properties)
Limitations
The call routing service has the following limitations:

A call coming from a SIP interface cannot be routed to another SIP interface. When that occurs,
the call automatically fails.


A call automatically fails if it is redirected to a route or hunt more than 10 times.







Maximum number of Routes: 40
The call properties Called Bearer Channel and Calling Bearer Channel are limited to ISDN
interfaces only.
Maximum number of Mapping Types: 40
Maximum number of Mapping Expressions: 100
Maximum number of Hunts: 40
Maximum number of Signaling Properties: 40
Maximum number of SIP Header Translations: 100
Maxium number of Call Properties Translations: 100
Regular Expressions
Some of the routing types described in “Routing Type” on page 434 require that you enter them following the
regular expression syntax. A regular expression is a string used to find and replace strings in other large
strings. The Mediatrix unit uses regular expressions to enter a value in several routing types, often by using
wildcard characters. These characters provide additional flexibility in designing call routing and decrease the
need for multiple entries in configuring number ranges.
The expression cannot begin by “^”, it is implicit in the expression. The following table shows some of the
wildcard characters that are supported:
Table 362: Regular Expressions Wildcards
Character
.
432
Description
Single-digit place holder. For instance, 555 .... matches any dialed number beginning with
555, plus at least four additional digits. Note that the number may be longer and still match.
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Table 362: Regular Expressions Wildcards (Continued)
Character
*
Description
Repeats the previous digit 0, 1, or more times. For instance, in the pattern:
1888*1
the pattern matches:
1881, 18881, 188881, 1888881
Note: If you are trying to handle the asterisk (*) as part of a dialed number, you must use \*.
[]
Range of digits.
•
A consecutive range is indicated with a hyphen (-), for instance, [5-7].
•
A nonconsecutive range is indicated without a delimiter, for instance, [58].
•
Both can be used in combination, for instance [5-79], which is the same as
[5679].
You may place a (^) symbol right after the opening bracket to indicate that the specified range
is an exclude list. For instance, [^01] specifies the same range as [2-9].
Note: The call router only supports single-digit ranges. You cannot specify the range of
numbers between 99 and 102 by using [99-102].
()
Indicates a pattern (also called group), for instance, 555(2525). It is used when replacing a
number in a mapping. See “Groups” on page 433 for more details.
?
Matches 0 or 1 occurrence of the previous item. For instance, 123?4 matches both 124 and
1234.
+
Repeats the previous digit one or more time. For instance 12+345 matches 12345, 122345,
etc. (but not 1345). If you use the + at the end of a number, it repeats the last number one or
more times. For instance: 12345+ matches, 12345, 123455, 1234555, etc.
|
Indicates a choice of matching expressions (OR).
The matching criterion implicitly matches from the beginning of the string, but not necessarily up to the end.
For instance, 123 will match the criterion 1, but it will not match the criterion 2.
If you want to match the whole string, you must end the criterion with “$”. For instance, 123 will not match the
criterion 1$ and will match the criterion 123$.
Note: You can use the “<undefined>” string if you want to match a property that is not defined.
You can also use the macro “local_ip_port“ to replace the properties by the local IP address and port of the
listening network of the SIP gateway used to send the INVITE.
Groups
A group is placed within parenthesis. It is used when replacing a string in a mapping. You can use up to nine
groups (defined by “\1” to “\9”) and matching is not case sensitive. “\0” represents the whole string. Lets say
for instance you have the following string:
9(123(45)6)
The following describes how the groups are replaced in a properties manipulation:
Table 363: Groups Replacement Example
Replacement
Dgw v2.0 Application
Result
\0
9123456
\1
123456
\2
45
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Chapter 44 - Call Router Configuration
Introduction
Table 363: Groups Replacement Example
Replacement
Result
\3
Groups can only be used with the following routing types:




Calling/Called E.164
Calling/Called Name
Calling/Called Host
Calling/Called URI
Routing Type
The following sub-sections list the available routing types of the call router and their supported values. The
routing types that offer choices use the choices as defined in the Q.931 standard. Q.931 is ISDN’s connection
control protocol, roughly comparable to TCP in the Internet protocol stack. The values may also be a special
tag, as described in “Special Tags” on page 440.
Table 364: Routing Types Locations
Routing Type
Location
E164
“Called / Calling E164” on page 435
Type of Number (TON)
“Called / Calling TON” on page 435
Numbering Plan Indicator (NPI)
“Called / Calling NPI” on page 435
Name
“Called / Calling Name” on page 435
Host
“Called / Calling Host” on page 436
URI
“Called / Calling URI” on page 436
Presentation Indicator (PI)
“Calling PI” on page 436
Screening Indicator (SI)
“Calling SI” on page 436
Information Transfer Capability (ITC) “Calling ITC” on page 436
Date and Time
“Date/Time” on page 437
Phone Context
“Called / Calling Phone Context” on page 438
SIP Username
“Called / Calling SIP Username” on page 438
Bearer Channel
“Called / Calling SIP Username” on page 438
Diverting Reason
“Last / Original Diverting Reason” on page 438
Diverting E.164
“Last / Original Diverting E.164” on page 438
Diverting Party Number Type
“Last / Original Diverting Party Number Type” on page 438
Diverting Public Type Of Number
“Last / Original Diverting Public Type Of Number” on page 439
Diverting Pivate Type Of Number
“Last / Original Diverting Private Type Of Number” on page 439
Diverting Number Presentation
“Last / Original Diverting Number Presentation” on page 439
SIP Privacy Type
“SIP Privacy Type” on page 439
Media5 recommends to carefully define the routing requirements and restrictions that apply to your installation
before starting the routing configuration. This will help you determine the types of routing you need. When this
is done, define the routes and mappings, as well as the hunts that you need to fulfil these requirements. You
may need several entries of the same type to achieve your goals.
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See also “Call Properties Parameters” on page 440 for a description of the parameters used by the various
routing types and interfaces of the call router.
Called / Calling E164
This is the Called/Calling Party Number. You can enter a regular expression (called/calling party E.164
number in the call setup message) as per “Regular Expressions” on page 432. Note that:

A PBX may insert or modify the calling party number. Sometimes there is no calling party
number at all. This all depends on the equipment you connect to the device.

The Mediatrix unit cannot filter the redirecting number information element of the SETUP
message because it does not support the “calling-Redir-E164” and “Calling-Redir-Reason”
routing properties criteria.
Called / Calling TON
Called or calling party type of number field in the ISDN setup message. The following values are available:
Table 365: Type of Number Values
Value
unknown
Description
Unknown number type.
international International number.
national
National number.
network
Network specific number used to indicate an administration or service number specific to the
serving network.
subscriber
Subscriber number.
abbreviated
Abbreviated number.
Note: The called type of number is set to international if the To username is an E.164 with the prefix “+”.
The calling type of number is set to international if the From username is an E.164 with the prefix “+”.
Called / Calling NPI
Called or calling party numbering plan indicator field in the ISDN setup message. The following values are
available:
Table 366: Numbering Plan Indicator Values
Value
Description
unknown
Unknown numbering plan.
isdn (E.164)
ISDN/Telephony numbering plan according to ITU-T Recommendation E.164.
data (X.121) Data numbering plan according to ITU-T Recommendation X.121.
telex (F.69)
Telex numbering plan according to ITU-T Recommendation F.69.
national
Numbering plan according to a national standard.
private
A private numbering plan.
Called / Calling Name
Calling and called party name (display name). This is the human-readable name of the calling or called party.
See “Regular Expressions” on page 432 for more details on how to enter a proper expression.
The Mediatrix unit does not support the sending of the calling name in the user-to-user information element.
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Called / Calling Host
IP address or domain name of the called or calling host in the following format:
Fqdn[:port]
If [:port] is missing, the call router uses the well-known port of the signalling protocol. Note that:

Incoming SIP calls use the calling party IP address property to store the IP address of the
remote SIP user agent. Other interfaces such as ISDN set the IP address to 0.0.0.0.
You can use a regular expression to enter an IP address or a range of IP addresses.
Called / Calling URI
Uniform Resource Identifier (URI) of:


the called party, e.g., the To-URI.
the originating VoIP peer, e.g., the From-URI of an incoming SIP call.
The URI follows the format described in RFC 3261.
Calling PI
Presentation indicator of the calling party number. The following values are available:
Table 367: Presentation Indicator Values
Value
Description
allowed
Presentation of the calling party number is allowed.
restricted
Presentation of the calling party number is restricted.
interworking The calling party number is not available due to interworking.
You may want to remove the calling party number when the user sets the presentation indicator to restricted.
To achieve this, route restricted calls to a mapping that sets the Calling E164 to an empty string.
Calling SI
Screening indicator of the calling party number. The following values are available:
Table 368: Screening Indicator Values
Value
Description
notscreened
The user provides the calling party number but the number is not screened by the network.
Thus the calling party possibly sends a number that it does not own.
passed
The calling party number is provided by the user and it passes screening.
failed
The calling party number is set by the user and verification of the number failed.
network
The originating network provides the number in the calling party number parameter.
You may want to remove the calling party number when it is not screened or screening failed. To do so, route
these calls to a mapping that sets the Calling E164 to an empty string. If you want to drop calls when the calling
party number is not screened or screening failed, use the Calling Si as criteria for the route.
Calling ITC
The information transfer capability field of the bearer capability information element in the ISDN setup
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message. The following values are available:
Table 369: Information Transfer Capability Values
Value
Description
speech
Voice terminals (telephones).
unrestricted
Unrestricted digital information (64 kbps).
restricted
Restricted digital information (64 kbps).
3.1Khz
Transparent 3.1 kHz audio channel.
udi-ta
Unrestricted digital information with tones/announcements.
Note: This was formerly transparent 7.1 kHz audio channel.
video
Video conference terminals.
The Mediatrix unit currently supports the following Information Transfer Capabilities when receiving calls to
and from the ISDN (named as in Q.931, 05/98):



Speech
Unrestricted Digital Information
3.1 kHz Audio
Those are respectively referenced as Speech, Unrestricted and 3.1 kHz in the call routing configuration.
When initiating calls towards the ISDN, the Mediatrix unit uses the calling ITC value if it is one of the three
listed above. If none is set, it uses 3.1 kHz Audio. If the calling ITC set by the call router is different from the
three listed above, the call is rejected.
Note: Terminals connected to analog extensions (e.g. of a PBX) do not supply information transfer
capability values in their call setup. The configuration of the analog port on the Terminal Adapter, NT or PBX
is thus responsible to insert this value. The configuration of this value is however often omitted or wrong.
The ITC value may therefore not be a reliable indication to differentiate between analogue speech, audio or
Fax Group 3 connections. Furthermore, calls from SIP interfaces do not differentiate between bearer
capabilities. They always set the information transfer capability property to 3.1Khz.
Date/Time
Day of week and time period and/or date and time period. The following are the accepted formats:
Table 370: Date/Time Accepted Formats
Format
Date/Time Period format
Week Day/Time Period format
Description
•
'DD.MM.YYYY/HH:MM:SS-DD.MM.YYYY/
HH:MM:SS'
•
'DD.MM.YYYY/HH:MM:SS-HH:MM:SS'
•
'DD.MM.YYYY-DD.MM.YYYY'
•
'DD.MM.YYYY'
•
'HH:MM:SS-HH:MM:SS'
•
'DDD'
•
'DDD,DDD...'
•
'DDD/HH:MM:SS-HH:MM:SS'
•
'DDD,DDD.../HH:MM:SS-HH:MM:SS'
DDD must be one of: SUN, MON, TUE, WED, THU, FRI,
SAT.
Many of the formats above can be concatenated to form one expression. They must be separated by |. For
instance: 25.12.2006 | SUN.
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Called / Calling Phone Context
This is a user parameter in a URI. For instance:
sip:1234;phone-context=1234@domain.com;user=phone
You can enter a regular expression (called/calling party phone context in the call setup message) as per
“Regular Expressions” on page 432.
Called / Calling SIP Username
Calling and called party SIP username. See “Regular Expressions” on page 432 for more details on how to
enter a proper expression.
Called / Calling Bearer Channel
Calling and called party bearer channel. See “Regular Expressions” on page 432 for more details on how to
enter a proper expression.
Last / Original Diverting Reason
This is the last or original diverting reason in ISDN setup and SIP INVITE messages. The following values are
available:
Table 371: Diverting Reason Values
Value
Description
cfb
Call Forward on Busy – Allowed.
cfu
Call Forward on Unavailable – Restricted
cfnr
Call Forward on No Answer – Interworking
unknown
unknown
Refer to “You can set the SIP transfer method when an endpoint is acting as the transferor in a blind transfer
scenario.” on page 314 to select the SIP method used to receive/send call diversion information in an INVITE.
Last / Original Diverting E.164
Last or original party number to which the call was being routed when the first diversion occurred. You can
enter a regular expression (called/calling party E.164 number in the call setup message) as per “Regular
Expressions” on page 432. Note that:

A PBX may insert or modify the calling party number. Sometimes there is no calling party
number at all. This all depends on the equipment you connect to the device.

The Mediatrix unit cannot filter the redirecting number information element of the SETUP
message because it does not support the “calling-Redir-E164” and “Calling-Redir-Reason”
routing properties criteria.
Last / Original Diverting Party Number Type
The following values are available:
Table 372: Diverting Party Number Type Values
Value
438
Description
unknown
Unknown number type.
public
Public number.
private
Private number.
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Last / Original Diverting Public Type Of Number
Diverting or original called number public type of number field in the ISDN Setup message. Used only when
the diverting or original called number type of number is 'public'. The following values are available:
Table 373: Diverting Public Type of Number Values
Value
unknown
Description
Unknown number type.
international International number.
national
National number.
networkspecific
Network specific number used to indicate an administration or service number specific to the
serving network.
subscriber
Subscriber number.
abbreviated
Abbreviated number.
Last / Original Diverting Private Type Of Number
Diverting or original called number private type of number field in the ISDN Setup message. Used when the
diverting or original called party number type is 'private'. The following values are available:
Table 374: Diverting Private Type of Number Values
Value
Description
unknown
Unknown.
leg2-reg
Leg2 reg.
leg1-reg
Leg1 reg.
pisnspecific
PISN Specific.
subscriber
Subscriber number.
abbreviated
Abbreviated number.
Last / Original Diverting Number Presentation
Diverting or original called number presentation. The following values are available:
Table 375: Diverting Presentation Values
Value
Description
allowed
Presentation of the party number is allowed.
restricted
Presentation of the party number is restricted.
interworking The party number is not available due to interworking.
restrictedaddress
Restricted address.
SIP Privacy Type
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Calling SIP privacy level of the call. The following values are available:
Table 376: SIP Privacy Values
Value
Description
disabled
No privacy is used.
none
Use P-Asserted Identity privacy.
id
Use P-Preferred Identity privacy.
Special Tags
You can use the following special tags as routing types values.
Table 377: Special Tags
Tag
Description
undefined
Matches if the property is not defined for the call.
default
Always matches. Generally used to set a default route if the previous criteria do not match.
Call Properties Parameters
The following sections describe the parameters used by the various call properties (routing types) and
interfaces of the call router.
Call Properties to SIP
This section describes the information the call router uses for the various SIP fields.
Table 378: Call Properties to SIP
SIP Field
To
440
Description
The Mediatrix unit uses the calling URI to populate the To field if not undefined.
Otherwise, the unit does the following:
•
Uses the called Name for the friendly name if not undefined.
•
Uses the called SipUsername for the user name if not empty or
undefined; otherwise, uses the called E164 for the username. If it is
empty or undefined, the Mediatrix unit rather uses the value defined
in the Default Username Value field of the SIP > Interop > SIP
Interop parameters as username (see “SIP Interop” on page 279 for
more details). The unit uses the called Phone Context for the user's
'phone-context' parameter if not empty. If a 'phone-context'
parameter is added, the URI parameter 'user' is also automatically
added. Its value is defined in the SIP URI User Parameter Value
field of the SIP > Interop > SIP Interop parameters. If empty, then
the value 'phone' is used
•
Uses the called Host for the host if not undefined, otherwise uses the
configured home domain proxy host.
•
Prefixes the user name with “+” and adds the URI parameter “user”
with the value “phone” if the called TON is “international”.
•
If there is no URI parameter “user” yet and the SIP URI User
Parameter Value field of the SIP > Interop > SIP Interop parameters
is not empty, then the parameter is added with the value defined by
the field.
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Table 378: Call Properties to SIP (Continued)
SIP Field
From
Dgw v2.0 Application
Description
The Mediatrix unit uses the called URI to populate the From field if not
undefined. Otherwise, the unit does the following:
•
Uses the calling Name for the friendly name if not undefined.
•
Uses the calling SipUsername for the user name if not empty or
undefined; otherwise, uses the calling E164 for the username. If it is
empty or undefined, the Mediatrix unit rather uses the value defined
in the Default Username Value field of the SIP > Interop > SIP
Interop parameters as username (see “SIP Interop” on page 279 for
more details).The unit uses the calling Phone Context for the user's
'phone-context' parameter if not empty. If a 'phone-context'
parameter is added, the URI parameter 'user' is also automatically
added. Its value is defined in the SIP URI User Parameter Value
field of the SIP > Interop > SIP Interop parameters. If empty, then
the value 'phone' is used.
•
Uses the calling Host for the host if not undefined, otherwise uses
the configured home domain proxy host.
•
Prefixes the user name with “+” and adds the URI parameter “user”
with the value “phone” if the calling TON is “international”.
•
If there is no URI parameter “user” yet and the SIP URI User
Parameter Value field of the SIP > Interop > SIP Interop parameters
is not empty, then the parameter is added with the value defined by
the field.
Request URI
The Mediatrix unit uses the same information as the To field.
Contact
The Mediatrix unit uses the same information as the From field, but with the
current IP address/port for the host.
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Table 378: Call Properties to SIP (Continued)
SIP Field
Diversion
Description
A Diversion header is added if the Last Diverting E.164 property is present and
not empty. This Diversion header is constructed as follows:
•
The username of the URI is set to the value of the Last Diverting
E.164 property.
•
The host of the URI is set to the configured home domain proxy host.
•
The reason field is set according to value of the Last Diverting
Reason property:
•
•
cfu: "unconditional"
•
cfb: "user-busy"
•
cfnr: "no-answer"
•
All other values or when undefined: "unknown'.
The field counter is set to the value of DivertingCounter if the
Original Diverting E.164 property is set to empty or undefined,
otherwise it is set to DivertingCounter -1.
A second Diversion header is added if the Last Diverting E.164 and Original
Diverting E.164 properties are present and not empty. This Diversion header is
constructed as follows:
•
The username of the URI is set to the value of the Original Diverting
E.164 property.
•
The host of the URI is set to the configured home domain proxy host.
•
The reason field is set according to the value of the Original
Diverting Reason property:
•
cfu: "unconditional"
•
cfb: "user-busy"
•
cfnr: "no-answer"
•
All other values or when undefined: "unknown'.
The field counter is set to 1.
SIP to Call Properties
This section describes the SIP information the call router uses for the various call properties.
Table 379: SIP to Call Properties
Property
442
SIP Information
Called URI
The URL of the To field.
Calling URI
The URL of the From field.
Called Name
The friendly name in the To field. The property is undefined if there is no
friendly name.
Calling Name
The friendly name in the From field. The property is undefined if there is no
friendly name.
Called E164
The user name of the Request-Uri field if the user name is a compatible E.164.
The prefix “+” and separator “-” are removed. The property is undefined if there
is no user name or if it is not compatible.
Calling E164
The user name of the From field if the user name is a compatible E.164. The
prefix “+” and separator “-” are removed. The property is undefined if there is no
user name or if it is not compatible.
Called Host
The host of the To field.
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Table 379: SIP to Call Properties (Continued)
Property
SIP Information
Calling Host
The host of the Contact field.
Called TON
Set to “international” if the To user name is an E.164 with the prefix “+”;
otherwise, the property is undefined.
Calling TON
Set to “international” if the From user name is an E.164 with the prefix “+”;
otherwise the property is undefined.
Called Phone Context
Set to the parameter “phone-context” of the user name of the To if the user
name is an E.164, otherwise the property is undefined.
Calling Phone Context
Set to the parameter “phone-context” of the user name of the From if the user
name is an E.164, otherwise the property is undefined.
Called SIP Username
Set to the username of the Request-Uri. Note that this does not include the
username parameter like the “phone-context”.
Calling SIP Username
Set to the username of the From. Note that this does not include the username
parameter like the “phone-context”.
Last Diverting Reason
If the INVITE contains at least one Diversion header, this value is set according
to the reason field value of the first Diversion header:
•
"user-busy": cfb
•
"unconditional":cfu
•
"no-answer": cfna
•
All other values: unknown
Otherwise, the property is undefined.
The reason field comparison is not case sensitive.
Original Diverting
Reason
If the INVITE contains more than one Diversion header, this value is set
according to the reason field value of the last Diversion header:
•
"user-busy": cfb
•
"unconditional":cfu
•
"no-answer": cfna
•
All other values: unknown
Otherwise, the property is undefined.
The reason field comparison is not case sensitive.
Last Diverting E.164
If the INVITE contains at least one Diversion header, this value is set to the
username of the URI (can be a SIP URI, SIPS URI or TEL URI) of the first
Diversion header converted into an E.164. It can be set to empty if there is no
username or if the username is not an E.164.
Otherwise, the property is undefined.
Original Diverting E.164
If the INVITE contains more than one Diversion header, this value is set to the
username of the URI (can be a SIP URI, SIPS URI or TEL URI) of the last
Diversion header converted into an E.164. It can be set to empty if there is no
username or if the username is not an E.164.
Otherwise, the property is undefined.
Diverting Counter
If the INVITE contains at least one Diversion header, this value is set to the
sum of the counter field of all Diversion headers. If a diversion header does not
contain the counter field, the value 1 is assumed for the header.
All others
The property is undefined.
Call Properties to ISDN
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This section describes the information the call router uses for the various ISDN information elements.
Table 380: Call Properties to ISDN
Information Element
Bearer Capabilities
Calling Party Number
Description
If valid, the calling ITC is used to fill the “information transfer capability” (octet 3
[5:1]). Otherwise, the ITC is set to “3.1 kHz audio”. If more than one bearer
capability information elements is provided in a prioritized list, they all receive
the same ITC. This information element is included in the SETUP message
only for outgoing calls.
Uses the calling E164 to fill the field “number digits” (octet 4).
Uses the calling TON to fill the field “type of number” (octet 3 [7:5]).
Uses the calling PI to fill the field “presentation indicator” (octet 3a [7:6).
Uses the calling SI to fill the field “screening indicator” (octet 3a [2:1]).
Uses the calling NPI to fill the field “numbering plan identification” (octet 3
[4:1]).
Called Party Number
Uses the called E164 to fill the field “number digits” (octet 4).
Uses the called TON to fill the field “type of number” (octet 3 [7:5]).
Uses the called NPI to fill the field “numbering plan identification” (octet 3 [4:1]).
Display
Uses the calling E164 to fill the field “display information” (octet 3).
Called Bearer Channel
The called bearer channel is used to select a specific ISDN bearer channel for
an outgoing ISDN call.
ISDN to Call Properties
This section describes the ISDN information the call router uses for the various call properties.
Table 381: ISDN to Call Properties
Property
444
ISDN Information
Calling Name
Field “display information” (octet 3) of the Display information element, if
included in the SETUP Q.931 message.
Called E164
Field “number digits” (octet 4) of the called party information element included
in the SETUP Q.931 message.
Calling E164
Field “number digits” (octet 4) of the calling party information element included
in the SETUP Q.931 message.
Called TON
Field “type of number” (octet 3 [7:5]) of the called party information element
included in the SETUP Q.931 message.
Calling TON
Field “type of number” (octet 3 [7:5]) of the calling party information element
included in the SETUP Q.931 message.
Calling PI
Field “presentation indicator” (octet 3a [7:6) of the calling party information
element included in the SETUP Q.931 message.
Calling SI
Field “screening indicator” (octet 3a [2:1]) of the calling party information
element included in the SETUP Q.931 message.
Calling ITC
Field “information transfer capability” (octet 3 [5:1]) of the bearer capability
information element included in the SETUP Q.931 message.
Called NPI
Field “numbering plan identification” (octet 3 [4:1]) of the called party
information element included in the SETUP Q.931 message.
Calling NPI
Field “numbering plan identification” (octet 3 [4:1]) of the calling party
information element included in the SETUP Q.931 message.
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Table 381: ISDN to Call Properties (Continued)
Property
ISDN Information
Calling Bearer Channel
Represents the ISDN bearer channel on which the ISDN call is received.
All others
The property is undefined.
Call Properties to FXS
This section describes the information the call router uses for the various call properties to FXS.
Table 382: Call Properties to FXS
Caller ID
Description
Number
If the PI property is present and not set to "allowed", the number is "P".
Otherwise, the number is set to the value of the E164 property (truncated to the
first 20 characters). See “Auto-Routing” on page 487 for details.
Name
If the PI property is present and not set to "allowed", the name is "Anonymous".
Otherwise, the name is set to the value of the Name property (truncated to the
first 50 characters). See “Auto-Routing” on page 487 for details.
FXS to Call Properties
This section describes the information the call router uses for the various FXS to call properties.
Table 383: FXS to Call Properties
Caller ID
Description
Calling E164
If the auto routing is enabled and the E164 field of the Call Router > Autorouting page is not empty (see “Auto-Routing” on page 487 for details), the
value of the E164 field. Otherwise, the property is not present.
Calling Name
If the auto routing is enabled and the Name field of the Call Router > Autorouting page is not empty (see “Auto-Routing” on page 487 for details), the
value of the Name field. Otherwise, the property is not present.
Calling SIP Username
If the auto routing is enabled and the SIP Username field of the Call Router >
Auto-routing page is not empty (see “Auto-Routing” on page 487 for details),
the value of the SIP Username field. Otherwise, the property is not present.
Called E164
For automatic calls, the E.164 defined in the Automatic Call Target field of the
Telephony > Services page (see “Automatic Call” on page 387 for more
details).
For other calls, the dialed digit after the transformation defined in the
Transformation field of the Allowed DTMF Map section (Telephony > DTMF
Maps page – see “Allowed DTMF Maps” on page 373 for more details).
Called Name
For automatic calls, the name specified in the Automatic Call Target field of the
Telephony > Services page (see “Automatic Call” on page 387 for more
details). The property is not present if the target address does not contain a
name.
For other calls, the property is not present.
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Table 383: FXS to Call Properties (Continued)
Caller ID
Called Host
Description
For automatic calls, the host specified in the the Automatic Call Target field of
the Telephony > Services page (see “Automatic Call” on page 387 for more
details). The property is not present if the target address does not contain a
host.
For other calls, the host defined in the Target field of the Allowed DTMF Map
section (Telephony > DTMF Maps page – see “Allowed DTMF Maps” on
page 373 for more details). The property is not present if the target host is not
configured for the matching DTMF map.
Call Properties to FXO
This section describes the information the call router uses for the various call properties to FXO.
Table 384: Call Properties to FXO
Caller ID
Dialled number
Description
The Called E164 property.
FXO to Call Properties
This section describes the information the call router uses for the various FXO to call properties.
Table 385: FXO to Call Properties
Caller ID
Calling E164
Description
If the caller ID is detected, the numbers provided by the caller ID.
If the auto routing is enabled and the E164 field of the Call Router > Autorouting page is not empty (see “Auto-Routing” on page 487 for details), the
value of the E164 field. Otherwise, the property is not present.
Calling Name
If the caller ID is detected, the name provided by the caller ID.
If the auto routing is enabled and the Name field of the Call Router > Autorouting page is not empty (see “Auto-Routing” on page 487 for details), the
value of the Name field. Otherwise, the property is not present.
Calling SIP Username
If the caller ID is detected, the property is not present.
If the auto routing is enabled and the SIP Username field of the Call Router >
Auto-routing page is not empty (see “Auto-Routing” on page 487 for details),
the value of the SIP Username field. Otherwise, the property is not present.
Called E164
For automatic calls, the E.164 defined in the Automatic Call Target field of the
Telephony > Services page (see “Automatic Call” on page 387 for more
details).
For other calls, the dialed digit after the transformation defined in the
Transformation field of the Allowed DTMF Map section (Telephony > DTMF
Maps page – see “Allowed DTMF Maps” on page 373 for more details).
Called Name
For automatic calls, the name specified in the Automatic Call Target field of the
Telephony > Services page (see “Automatic Call” on page 387 for more
details). The property is not present if the target address does not contain a
name.
For other calls, the property is not present.
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Table 385: FXO to Call Properties (Continued)
Caller ID
Called Host
Description
For automatic calls, the host specified in the the Automatic Call Target field of
the Telephony > Services page (see “Automatic Call” on page 387 for more
details). The property is not present if the target address does not contain a
host.
For other calls, the host defined in the Target field of the Allowed DTMF Map
section (Telephony > DTMF Maps page – see “Allowed DTMF Maps” on
page 373 for more details). The property is not present if the target host is not
configured for the matching DTMF map.
SIP/ISDN Call Default Values
When performing a call from SIP to ISDN or ISDN to SIP, some ISDN informations are missing from the SIP
packet. The Dgw v2.0 Application sets the following default values when the information is missing. You
cannot filter on these default values, but you can filter with the “<undefined>” or “<default>” values.
Table 386: SIP/ISDN Calls Default Values
Parameter
Default Value
SIP to ISDN Calls
TON (calling)
unknown
TON (called)
unknown
NPI (calling and called) unknown
SI (calling)
User-side: not-screened
Network-side: network
ITC (calling)
PI (calling)
3.1 kHz audio
1. When the Calling Party Number E.164 is missing: interworking. In this
case, this value overrides any value set by the call router.
2. When CLIR is enabled (user-side only): restricted. In this case, this value
overrides any value set by the call router.
3. All other cases: allowed. This is the default value if the two cases above
do not apply and no value has been set by the call router.
ISDN to SIP Calls
SI (calling)
Network-side: The SI in the incoming Calling Party information element is ignored
and replaced by one of the following:
1. No calling IA5 digits received: network.
2. NPI is not “unknown” nor “ISDN telephony”: network.
3. TON is not “international” nor “national”: network, called IA5 digits are
discarded.
4. PI is set to “interworking”: network.
5. Otherwise: passed.
User-side: not-screened.
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Table 386: SIP/ISDN Calls Default Values (Continued)
Parameter
PI (calling)
Default Value
Network-side:
1. CLIR enabled: restricted. The PI is set to restricted no matter if a PI is
present in the incoming Calling Party IE.
2. CLIR disabled, no IA5 digits provided: interworking.
3. CLIR disabled, IA5 digits provided: allowed.
User-side:
1. CLIR disabled, no IA5 digits provided: interworking.
2. CLIR disabled, IA5 digits provided: allowed.
ITC (calling)
Must be provided in the incoming Bearer Capabilities information element
provided by the ISDN peer that initiated the call. There is no default value, the call
should be rejected if missing.
TON (called)
The Called TON must be provided by the ISDN peer that initiated the call.
TON (calling)
unknown
NPI (called
The Called NPI must be provided by the ISDN peer that initiated the call.
NPI (calling)
unknown
Note that the calling PI, SI, TON and NPI are present in Calling Party information elements in SETUP
messages sent by the network-side only when CLIP is enabled. They should always be present in messages
sent by the user-side. See “Chapter 23 - ISDN Configuration” on page 149 for more details on CLIP.
Call Routing Status
The routes, mappings, and hunts currently in use, as well as the available interfaces, are displayed in the Call
Router > Status page.
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Figure 199: Call Router – Status Web Page
Routes
The routing table contains one or more routes. These routes forward an incoming or outgoing call to another
route, interface, or hunt based on a specific call property such as the called party number. It may also use a
mapping to modify the call setup message of a call and a signalling property to modify the behaviour of the
call at the SIP protocol level.
Once the call router finds a route that matches, it does not check the other routes, even if some of them may
still match. The routes sequence is thus very important. The call router follows the routing table rows (routes)
as they are entered in the web interface. If you want the call router to try to match one row before another one,
you must put that row first.
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When a call arrives, the call router proceeds as follows:
1.
It examines the call property as specified with the routes.
To select a route, the call must match all three of the Source, Properties Criteria, and Expression
Criteria parameters.
2.
It selects the first matching route in the list of routes.
3.
It routes the call to the specified destination interface, hunt, or route.
Note: You can revert back to the configuration displayed in the Call Router > Status web page at any time
by clicking the Rollback button at the bottom of the page. All modified settings in the Call Router > Route
Config page will be lost.
You can add up to 40 routes.
Creating/Editing a Route
The web interface allows you to create a route or modify the parameters of an existing one.
 To create or edit a route:
1.
In the web interface, click the Call Router link, then the Route Config sub-link.
Figure 200: Call Router – Route Config Web Page
2
3
2.
Locate the Route section.
3.
Do one of the following:
•
If you want to add a route before an existing entry, locate the proper row in the table
and click
of this row.
•
If you want to add a route at the end of the existing rows, click
the Route section.
at the bottom right of
•
If you want to edit an existing route, locate the proper row in the table and click
.
This brings you to the Configure Route panel.
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Figure 201: Configure Route Panel
4
5
6
7
8
4.
9
Enter one or more sources to compare with the call and match in order to select the route in the
Source field.
You can use the Suggestion column’s drop-down menu to select between suggested values, if any.
A source may be:
•
route-name: The call uses the route name.
•
sip-name: The call comes from the SIP interface name.
•
isdn-name: The call comes from the ISDN interface name.
•
r2-name: The call destination is set to the R2 interface name.
•
e&m-name: The call comes from the E&M interface name.
•
fxs-name: The call destination is set to the FXS interface name.
•
fxo-name: The call destination is set to the FXO interface name.
If you want to use multiple sources, you must separate them by commas.
For instance, if you want to route calls that come from the SIP interface “default”, enter the following
value:
sip-default
If you want to route calls that come from the SIP interfaces “default” and “other”, enter the following
value:
sip-default,sip-other
Keep in mind that to select a route, the call must match all three of the Source, Properties Criteria,
and Expression Criteria parameters.
Note: When using endpoint gateways, SIP interface names are composed of both the gateway name and
a username; for example, a SIP source on an endpoint gateway may be: sip-default/5551212. When using
trunk gateways, SIP interface names are based on the gateway name only.
5.
Select a call property to compare with the call and match in order to select the route in the Properties
Criteria drop-down menu.
The call router offers several different routing types. Each type specifies which call property the call
router examines.
Table 387: Routing Types
Type
Dgw v2.0 Application
Description
Called E164
Routes calls based on the called party E.164 number.
Calling E164
Routes calls based on the calling party E.164 number.
Called TON
Routes calls based on the called party type of number.
Calling TON
Routes calls based on the calling party type of number.
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Table 387: Routing Types (Continued)
Type
Description
Called NPI
Routes calls based on the called party numbering plan indicator.
Calling NPI
Routes calls based on the calling party numbering plan indicator.
Called Name
Routes calls based on the display name of the called party.
Calling Name Routes calls based on the display name of the calling party.
Called Host
Routes calls based on the signalling IP address or domain name.
Calling Host
Routes calls based on the signalling IP address or domain name.
Called URI
Routes calls based on the To-URI.
Calling URI
Routes calls based on the From-URI.
Calling PI
Routes calls based on the presentation indicator.
Calling SI
Routes calls based on the screening indicator.
Calling ITC
Routes calls based on the information transfer capability.
Date/Time
Routes calls based on the date and/or time the call arrived at the call router. A
link called Time criteria editor appears on the right of the Expression criteria
field. Use it to easily configure the Date/Time type.
Called Phone
Context
Routes calls based on the called party phone context.
Calling
Phone
Context
Routes calls based on the calling party phone context.
Called SIP
Username
Routes calls based on the called party SIP username.
Calling SIP
Username
Routes calls based on the calling SIP username.
Called
Bearer
Channel
Routes calls based on the called bearer channel properties.
Calling
Bearer
Channel
Routes calls based on the calling bearer channel properties.
Calling SIP
Privacy
Routes calls based on the calling SIP privacy properties.
Keep in mind that to select a route, the call must match all three of the Source, Properties Criteria,
and Expression Criteria parameters.
6.
Enter the expression (related to the call properties selected in the previous step) to compare with
the call and match in order to select the route in the Expression Criteria field.
You can use the Suggestion column’s drop-down menu to select between suggested values, if any.
See “Routing Type” on page 434 for a list of available values for each call property.
For instance, if the property is Calling TON, you could instruct the call router to look for the following
expression:
international
If you have selected the Date/Time property in the above step, you can click the Time criteria
editor link and use the editor to easily configure the Date/Time parameters.
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Figure 202: Date/Time Criteria Editor (Day Time)
•
Select between the Day-Time or Time-Period settings in the Select Criteria Type dropdown menu. If you select Time-Period, the editor changes as follows:
Figure 203: Date/Time Criteria Editor (Time Period)
•
Select or enter the parameters you want, then click Add to List . If a parameter is
invalid (for instance, the end date is inferior to the start date), it is displayed in red in the
Time Criteria List field.
•
To remove an existing parameter, select it in the Time Criteria List field, then click
Remove Selected .
•
To update an existing parameter, select it in the Time Criteria List field, then click
Update Selected .
•
To remove all parameters, click Clear Parameters .
•
When done, click Save .
Keep in mind that to select a route, the call must match all three of the Source, Properties Criteria,
and Expression Criteria parameters.
7.
If applicable, enter the name of mappings to apply to the call in the Mappings field.
You can enter more than one mapping by separating them with commas. These mappings are
executed in sequential order.
You can use the Suggestion column’s drop-down menu to select an existing mapping, if any.
The manipulations are executed before sending the call to the new destination. See “Mappings” on
page 455 for more details.
If you leave this field empty, no mapping is required.
8.
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Select the call signalling property of the route used to modify the behaviour of the call at the SIP
protocol level in the Call Signaling drop-down menu.
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You must set call signaling properties as defined in “Signalling Properties” on page 465. You can
use the Suggestion column’s drop-down menu to select between existing properties, if any.
9.
Select the destination of the call when it matches in the Destination field.
You can use the Suggestion column’s drop-down menu to select between suggested values, if any.
The destination can be:
•
route-name: The call destination is set to the route name.
•
hunt-name: The call destination is set to the hunt name.
•
sip-name: The call destination is set to the SIP interface name.
•
isdn-name: The call destination is set to the ISDN interface name.
•
r2-name: The call destination is set to the R2 interface name.
•
e&m-name: The call destination is set to the E&M interface name.
•
fxs-name: The call destination is set to the FXS interface name.
•
fxo-name: The call destination is set to the FXO interface name.
•
SipRedirect-name: When the Route source is a SIP interface, incoming SIP Invites are
replied with a 302 'Moved Temporarily' SIP response. See “SIP Redirects” on page 483
or more details.
For instance, if you want to route calls to the hunt “CallCenter”, enter the following:
hunt-CallCenter
Note: When using endpoint gateways, SIP interface names are composed of both the gateway name and
a username; for example, a SIP source on an endpoint gateway may be: sip-default/5551212. When using
trunk gateways, SIP interface names are based on the gateway name only.
10.
Click Save.
This brings you back to the main Call Router > Route Config web page.
You can see a yellow Yes in the Config Modified section at the top of the window. It warns you that
the configuration has been modified but not applied (i.e., the Call Router > Status differs from the
Call Router > Route Config). The Route Config sub-menu is a working area where you build up a
Call Router configuration. While you work in this area, the configured parameters are saved but not
applied (i.e., they are not used to process incoming calls). The yellow Yes flag warns you that the
configuration has been modified but is not applied.
11.
Click Save to enable the route.
The current routes applied are displayed in the Call Router > Status web page. You can also see
that the yellow Config Modified Yes flag is cleared.
Examples
The following are some examples of routes:
Figure 204: Routes Examples
Moving a Route
Once the call router finds a routing entry that matches, it does not check the other entries, even if some of
them may still match. The routes sequence is thus very important. The call router follows the routing table rows
as they are entered in the web interface. If you want the call router to try to match one row before another one,
you must put that row first.
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 To move a routing entry up or down:
1.
Either click
or
of the row you want to move until the entry is properly located.
2.
Click Save to update the Call Router > Status web page.
Deleting a Route
You can delete a routing row from the table in the web interface.
 To delete a routing entry:
1.
Click
of the row you want to delete.
2.
Click Save to update the Call Router > Status web page.
Mappings
Mapping entries modify the call setup message of a call. They thus influence the routing decision and/or the
setup message leaving the call router. They are specifically called within a route.
Like the routing table, the mapping table finds the first matching entry. It then executes it by manipulating a
call property. A mapping always examines one call property and changes another property.
The call router executes all mapping entries that match by following the mapping table rows as they are
entered in the web interface. If you want the call router to try to match one row before another one, you must
put that row first.
The mapping may work with three types of call properties:



calling party properties
called party properties
generic properties
Generic properties are used for call properties that apply to both calling and called parties.
The web interface mapping configuration is separated in two parts: Mapping Type and Mapping Expression.
You must properly configure both parts for the mapping to work as required.
When a call arrives at the mapping table, the call router proceeds as follows:
1.
It examines the call property as specified in the Criteria (input) value of the Mapping Type part.
2.
It selects the first matching entry.
3.
It replaces the property specified in the Transformation (output) value of the Mapping Expression
part with the value of the selected entry.
Note: You can revert back to the configuration displayed in the Call Router > Status web page at any time
by clicking the Rollback button at the bottom of the page. All modified settings in the Call Router > Route
Config page will be lost.
Creating/Editing a Mapping Type
The Mapping Type part allows you to define the input call property to match and to define which call property
to change. The mapping type then uses one or more corresponding mapping expressions that you can define
in “Creating/Editing a Mapping Expression” on page 457.
You can add up to 40 Mapping Types.
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 To create or edit a mapping type:
1.
In the web interface, click the Call Router link, then the Route Config sub-link.
Figure 205: Call Router – Route Config Web Page
2
3
2.
Locate the Mapping Type section.
3.
Do one of the following:
•
If you want to add a mapping type entry before an existing entry, locate the proper row
in the table and click
of this row.
•
If you want to add a mapping type entry at the end of the existing rows, click
bottom right of the Mapping Type section.
•
If you want to edit an existing entry, locate the proper row in the table and click
at the
.
This brings you to the Configure Mapping Type panel.
Figure 206: Configure Mapping Type Panel
4
5
6
4.
Enter the name of the mapping in the Name field.
This is the name used in a route when calling a mapping. It must be unique. Media5 suggests to
use the type as part of the name for ease of identification.
There must be at least one corresponding mapping expression in the Mapping Expression table
with the exact same name. See “Creating/Editing a Mapping Expression” on page 457 for more
details.
456
5.
Select the input call property to compare with the call and match in order to select the mapping in
the Criteria drop-down menu.
6.
Select the call property to transform in the Transformation drop-down menu.
7.
Do one of the following:
•
Click Save to go back to the main Call Router > Route Config web page. You can now
define a corresponding mapping expression.
•
Click Save and Insert Expression to directly access the proper mapping expression
dialog.
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Creating/Editing a Mapping Expression
The Mapping Expression part defines the actual transformation to apply to the corresponding mapping type.
Each mapping expression must match a mapping type as defined in “Creating/Editing a Mapping Type” on
page 455.
You can add up to 100 Mapping Expressions.
 To create or edit a mapping expression:
1.
In the web interface, click the Call Router link, then the Route Config sub-link.
Figure 207: Call Router – Route Config Web Page
2
3
2.
Locate the Mapping Expression section.
3.
Do one of the following:
•
If you want to add a mapping expression entry before an existing entry, locate the
proper row in the table and click
of this row.
•
If you want to add a mapping expression entry at the end of the existing rows, click
at the bottom right of the Mapping Expression section.
•
If you want to edit an existing entry, locate the proper row in the table and click
.
This brings you to the Configure Mapping Expression panel.
Figure 208: Configure Mapping Expression Panel
4
5
6
7
4.
Enter the name of the mapping expression in the Name field.
This name must match a mapping type as defined in “Creating/Editing a Mapping Type” on
page 455. You can use the Suggestion column’s drop-down menu to select an existing mapping
type. When a name matches a mapping type, its type is displayed in the Type row as follows:
input type to output type
You can define several mapping expressions with the same name. In that case, the first row
matching the call is used. The rows are used in ascending order.
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5.
Mappings
Enter the expression (related to this specific input type) to compare with the call and match in order
to select the mapping in the Criteria field.
This string differs depending on the input type selected in the Mapping Type part (Criteria dropdown menu). For instance, if your input type is Calling TON, you could instruct the call router to look
for the following expression:
international
You can use the Suggestion column’s drop-down menu to select between suggested values, if any.
See “Routing Type” on page 434 for a list of available transformation values.
Table 388: Input Type Criteria
Input Type
Criteria
None
No criteria, always matches.
E164
If the Transformation value of the Mapping Type part is also a generic property,
this is applied to both this Calling E164 and Called E164 property.
Called E164
Selects an entry based on the called party E.164 number. You can use wildcards
to summarize entries as per “Called / Calling E164” on page 435.
Calling E164
Selects an entry based on the calling party E.164 number. You can use
wildcards to summarize entries as per “Called / Calling E164” on page 435.
Name
If the Transformation value of the Mapping Type part is also a generic property,
this is applied to both this Calling Name and Called Name property.
Called Name
Selects an entry based on the display name of the called party. You can use
wildcards to summarize entries as per “Called / Calling Name” on page 435.
Calling Name Selects an entry based on the display name of the calling party. You can use
wildcards to summarize entries as per “Called / Calling Name” on page 435.
458
TON
If the Transformation value of the Mapping Type part is also a generic property,
this is applied to both this Calling TON and Called TON property.
Called TON
Selects an entry based on the called party type of number as per “Called /
Calling TON” on page 435.
Calling TON
Selects an entry based on the calling party type of number as per “Called /
Calling TON” on page 435.
NPI
If the Transformation value of the Mapping Type part is also a generic property,
this is applied to both this Calling NPI and Called NPI property.
Called NPI
Selects an entry based on the called party numbering plan indicator as per
“Called / Calling NPI” on page 435.
Calling NPI
Selects an entry based on the calling party numbering plan indicator as per
“Called / Calling NPI” on page 435.
Host
If the Transformation value of the Mapping Type part is also a generic property,
this is applied to both this Calling Host and Called Host property.
Called Host
Selects an entry based on the remote signalling IP address or domain name of
the destination VoIP peer. You can use wildcards to summarize entries as per
“Called / Calling Host” on page 436.
Calling Host
Selects an entry based on the remote signalling IP address or domain name of
the originating VoIP peer. You can use wildcards to summarize entries as per
“Called / Calling Host” on page 436.
Calling PI
Selects an entry based on the presentation indicator as per “Calling PI” on
page 436.
Calling SI
Selects an entry based on the screening indicator as per “Calling SI” on
page 436.
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Table 388: Input Type Criteria (Continued)
Input Type
Criteria
Calling ITC
Selects an entry based on the information transfer capability as per “Calling ITC”
on page 436.
URI
If the Transformation value of the Mapping Type part is also a generic property,
this is applied to both this Calling URI and Called URI property.
Called URI
Selects an entry based on the called SIP URI properties. You can use wildcards
to summarize entries as per “Called / Calling URI” on page 436.
Calling URI
Selects an entry based on the calling SIP URI properties. You can use wildcards
to summarize entries as per “Called / Calling URI” on page 436.
Date/Time
Selects an entry based on the date and/or time the call arrived at the call router
as per “Date/Time” on page 437.
Phone
Context
Selects an entry based on the called or calling phone context properties as per
“Called / Calling Phone Context” on page 438.
Called Phone
Context
Selects an entry based on the called phone context properties as per “Called /
Calling Phone Context” on page 438.
Calling
Phone
Context
Selects an entry based on the calling phone context properties as per “Called /
Calling Phone Context” on page 438.
SIP
Username
Selects an entry based on the called or calling SIP username properties as per
“Called / Calling SIP Username” on page 438.
Called SIP
Username
Selects an entry based on the called SIP username properties as per “Called /
Calling SIP Username” on page 438.
Calling SIP
Username
Selects an entry based on the calling SIP username properties as per “Called /
Calling SIP Username” on page 438.
Last
Diverting
Reason
Selects an entry based on the last diverting reason properties as per “Last /
Original Diverting Reason” on page 438.
Last
Diverting
E164
Selects an entry based on the last diverting E.164 properties as per “Last /
Original Diverting E.164” on page 438.
Last
Selects an entry based on the party number type of the last diverting number
Diverting
properties as per “Last / Original Diverting Party Number Type” on page 438.
Party
Number Type
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Last
Diverting
Public Type
Of Number
Selects an entry based on the public type of number of the last diverting number
properties as per “Last / Original Diverting Public Type Of Number” on page 439.
Last
Diverting
Private Type
Of Number
Selects an entry based on the private type of numbekr of the last diverting
number properties as per “Last / Original Diverting Private Type Of Number” on
page 439.
Last
Diverting
Number
Presentation
Selects an entry based on the presentation of the last diverting number
properties as per “Last / Original Diverting Number Presentation” on page 439.
OriginalDiver
tingReason
Selects an entry based on the original diverting reason properties as per “Last /
Original Diverting Reason” on page 438.
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Table 388: Input Type Criteria (Continued)
Input Type
OriginalDiver
tingE164
Criteria
Selects an entry based on the original diverting E.164 properties as per “Last /
Original Diverting E.164” on page 438.
Original
Selects an entry based on the party number type of the original diverting number
Diverting
properties as per “Last / Original Diverting Party Number Type” on page 438.
Party
Number Type
Original
Diverting
Public Type
Of Number
Selects an entry based on the public type of number of the original diverting
number properties as per “Last / Original Diverting Public Type Of Number” on
page 439.
Called
Bearer
Channel
Selects an entry based on the called bearer channel properties as per “Called /
Calling SIP Username” on page 438.
Calling
Bearer
Channel
Selects an entry based on the calling bearer channel properties as per “Called /
Calling SIP Username” on page 438.
Calling SIP
Privacyl
Selects an entry based on the calling SIP privacy properties as per “SIP Privacy
Type” on page 439.
If you are editing a Date/Time property, you can click the Time criteria editor link and use the editor
to easily configure the Date/Time parameters.
Figure 209: Date/Time Criteria Editor (Day Time)
•
460
Select between the Day-Time or Time-Period settings in the Select Criteria Type dropdown menu. If you select Time-Period, the editor changes as follows:
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Figure 210: Date/Time Criteria Editor (Time Period)
6.
•
Select or enter the parameters you want, then click Add to List . If a parameter is
invalid (for instance, the end date is inferior to the start date), it is displayed in red in the
Time Criteria List field.
•
To remove an existing parameter, select it in the Time Criteria List field, then click
Remove Selected.
•
To update an existing parameter, select it in the Time Criteria List field, then click
Update Selected.
•
To remove all parameters, click Clear Parameters.
•
When done, click Save.
Enter the transformation (related to this specific output type) to apply in the Transformation field.
You can use the Suggestion column’s drop-down menu to select between suggested values, if any.
If the transformation is to replace part of an expression, it can use the matched group of the criteria.
“\0” will be replaced by the whole criteria capability and “\1” to “\9” by the matched group. See
“Groups” on page 433 for more details.
See “Routing Type” on page 434 for a list of available transformation values.
Table 389: Output Type Transformation
Output Type
Transformation
None
No transformation is applied.
E164
If the Criteria value of the Mapping Type part is also a generic property, this is
applied to both the Calling E164 and Called E164 properties.
Called E164
Modifies the called party E.164 number as per “Called / Calling E164” on
page 435.
Calling E164
Modifies the calling party E.164 number as per “Called / Calling E164” on
page 435.
Name
If the Criteria value of the Mapping Type part is also a generic property, this is
applied to both the Calling Name and Called Name properties.
Called Name
Sets the display name of the called party as per “Called / Calling Name” on
page 435.
Calling Name Sets the display name of the calling party as per “Called / Calling Name” on
page 435.
Dgw v2.0 Application
TON
If the Criteria value of the Mapping Type part is also a generic property, this is
applied to both the Calling TON and Called TON properties.
Called TON
Sets the called party type of number as per “Called / Calling TON” on page 435.
Calling TON
Sets the calling party type of number as per “Called / Calling TON” on page 435.
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Table 389: Output Type Transformation (Continued)
Output Type
Transformation
NPI
If the Criteria value of the Mapping Type part is also a generic property, this is
applied to both the Calling NPI and Called NPI properties.
Called NPI
Sets the called party numbering plan indicator as per “Called / Calling NPI” on
page 435.
Calling NPI
Sets the calling party numbering plan indicator as per “Called / Calling NPI” on
page 435.
Host
If the Criteria value of the Mapping Type part is also a generic property, this is
applied to both the Calling Host and Called Host properties.
Called Host
Sets the remote IP address or domain name of the destination VoIP peer as per
“Called / Calling Host” on page 436.
Calling Host
Sets the remote IP address or domain name of the originating VoIP peer as per
“Called / Calling Host” on page 436.
Calling PI
Sets the presentation indicator as per “Calling PI” on page 436.
Calling SI
Sets the screening indicator as per “Calling SI” on page 436.
Calling ITC
Sets the information transfer capability as per “Calling ITC” on page 436.
URI
If the Criteria value of the Mapping Type part is also a generic property, this is
applied to both the Calling URI and Called URI properties.
Called URI
Sets the called URI as per “Called / Calling URI” on page 436.
Calling URI
Sets the calling URI as per “Called / Calling URI” on page 436.
Phone
Context
If the Criteria value of the Mapping Type part is also a generic property, this is
applied to both the Calling Phone Context and Called Phone Context properties.
Called Phone
Context
Sets the called Phone Context as per “Called / Calling Phone Context” on
page 438.
Calling
Phone
Context
Sets the calling Phone Context as per “Called / Calling Phone Context” on
page 438.
SIP
Username
If the Criteria value of the Mapping Type part is also a generic property, this is
applied to both the Calling SIP Username and Called SIP Username properties.
Called SIP
Username
Sets the called SIP Username as per “Called / Calling SIP Username” on
page 438.
Calling SIP
Username
Sets the calling SIP Username as per “Called / Calling SIP Username” on
page 438.
Last
Diverting
Reason
Sets the last diverting reason properties as per “Last / Original Diverting
Reason” on page 438.
Last
Diverting
E164
Sets the last diverting E.164 properties as per “Last / Original Diverting E.164”
on page 438.
Sets the party number type of the last diverting number properties as per “Last /
Last
Diverting
Original Diverting Party Number Type” on page 438.
Party
Number Type
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Table 389: Output Type Transformation (Continued)
Output Type
Transformation
Last
Diverting
Public Type
Of Number
Sets the public type of number of the last diverting number properties as per
“Last / Original Diverting Public Type Of Number” on page 439.
Last
Diverting
Private Type
Of Number
Sets the private type of number of the last diverting number properties as per
“Last / Original Diverting Private Type Of Number” on page 439.
Last
Diverting
Number
Presentation
Sets the presentation of the last diverting number properties as per “Last /
Original Diverting Number Presentation” on page 439.
Original
Diverting
Reason
Sets the original diverting reason properties as per “Last / Original Diverting
Reason” on page 438.
Original
Diverting
E164
Sets the original diverting E.164 properties as per “Last / Original Diverting
E.164” on page 438.
Original
Sets the party number type of the original diverting number properties as per
Diverting
“Last / Original Diverting Party Number Type” on page 438.
Party
Number Type
Original
Diverting
Public Type
Of Number
Sets the public type of number of the original diverting number properties as per
“Last / Original Diverting Public Type Of Number” on page 439.
Original
Diverting
Private Type
Of Number
Sets the private type of number of the original diverting number properties as per
“Last / Original Diverting Private Type Of Number” on page 439.
Original
Diverting
Number
Presentation
Sets the Presentation of the original diverting number properties as per “Last /
Original Diverting Number Presentation” on page 439.
Called
Bearer
Channel
Sets the called bearer channel properties as per “Called / Calling SIP
Username” on page 438.
Calling
Bearer
Channel
Sets the calling bearer channel properties as per “Called / Calling SIP
Username” on page 438.
Debug
Reserved for debug configuration.
You cannot use Date/Time as an output type transformation.
7.
If applicable, enter the name of one or more subsequent mappings to execute in the Sub Mappings
field.
You can enter more than one mapping by separating them with commas. The mappings are
executed in sequential order.
You can use the Suggestion column’s drop-down menu to select between existing values, if any.
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You may want to send the result of the first mapping to another one. Once the subsequent mapping
is finished, the call router continues to check the mapping entries for matching entries. For instance,
if the call router is checking the fourth mapping entry and that entry uses subsequent mapping, the
call router executes the subsequent mapping, then resumes checking the fifth mapping entry, and
so on.
The maximal number of subsequent interleaved mapping is 3.
8.
Do one of the following:
•
Click Save to go back to the main Call Router > Route Config web page.
You can see a yellow Yes in the Config Modified section at the top of the window. It
warns you that the configuration has been modified but not applied (i.e., the Call Router
> Status differs from the Call Router > Route Config). The Route Config sub-menu is a
working area where you build up a Call Router configuration. While you work in this area,
the configured parameters are saved but not applied (i.e., they are not used to process
incoming calls). The yellow Yes flag warns you that the configuration has been modified
but is not applied.
•
9.
Click the Save and Insert Expression button to create another expression for the
same type.
Click Save to enable the mapping entry.
The current mappings applied are displayed in the Call Router > Status web page. You can also
see that the yellow Config Modified Yes flag is cleared.
Examples
The following are some examples of mappings:
Figure 211: Mappings Examples
Moving a Mapping Type or Expression Row
The mapping entries sequence is very important. The call router follows the mapping table rows as they are
entered in the web interface. If you want the call router to try to match one row before another one, you must
put that row first.
 To move a mapping entry up or down:
1.
In the Mapping Type or Mapping Expression table, either click
move until the entry is properly located.
2.
Click Save to update the Call Router > Status web page.
or
of the row you want to
Deleting a Mapping Type or Expression Row
You can delete a mapping row from the Mapping Type or Mapping Expression table in the web interface.
 To delete a mapping entry:
464
1.
Click
of the row you want to delete.
2.
Click Save to update the Call Router > Status web page.
Dgw v2.0 Application
Signalling Properties
Software Configuration Guide
Signalling Properties
Call signalling specifies how to set up a call to the destination Mediatrix unit or 3rd party equipment. Call
signalling properties are assigned to a route and used to modify the behaviour of the call at the SIP protocol
level.
Signaling Properties are applied after mappings rules.
Like the routing table, the signalling properties table finds the first matching entry. It then executes it by
modifying the behaviour of the call.
Note: You can revert back to the configuration displayed in the Call Router > Status web page at any time
by clicking the Rollback button at the bottom of the page. All modified settings in the Call Router > Route
Config page will be lost.
You can add up to 40 Signalling Properties.
Creating/Editing a Signalling Property
The web interface allows you to create a signalling property or modify the parameters of an existing one. The
signalling properties are called from a route as described in “Routes” on page 449.
 To create or edit a signalling property:
1.
In the web interface, click the Call Router link, then the Route Config sub-link.
Figure 212: Call Router – Route Config Web Page
2
3
2.
Locate the Signaling Properties section.
3.
Do one of the following:
•
If you want to add a signalling property entry before an existing entry, locate the proper
row in the table and click
of this row.
•
If you want to add a signalling property entry at the end of existing rows, click
bottom right of the Signaling Properties section.
•
If you want to edit an existing entry, locate the proper row in the table and click
at the
.
This brings you to the Configure Signaling Properties panel.
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Chapter 44 - Call Router Configuration
Signalling Properties
Figure 213: Configure Signaling Properties Panel
4
6
5
7
8
10
9
11
12
4.
Enter the name of the signalling property in the Name field.
The name must be unique. It will be used in routes to call a specific signalling property as described
in “Routes” on page 449.
5.
Select whether or not the early connect feature is enabled in the Early Connect drop-down menu.
When early connect is enabled, the SIP call is connected by sending a 200 OK message instead of
a 183 Session Progress message with early media, if the called party answers the call. It allows
interoperability with units that do not support the 183 Session Progress with SDP message.
When early connect is disabled, call progress tones or announcements are transmitted in the early
SIP dialog.
6.
Select whether or not the early disconnect feature is enabled in the Early Disconnect drop-down
menu.
This feature is useful to avoid hearing the end of call tone when the far end party terminates the call
during a conference.
When early disconnect is:
•
enabled, the SIP BYE message is sent upon receiving the ISDN “Disconnect” signal.
•
disabled, the SIP BYE message is sent upon receiving the ISDN “Call release” signal.
If early disconnect is enabled but no ISDN “Disconnect” message is received, the SIP BYE
message is sent upon receiving an ISDN “Call release” signal as if the early disconnect was
disabled.
7.
Define the SIP messages destination (where an INVITE is sent) in the Destination Host field.
It can override the Called Host property set by a mapping rule because signalling properties are
applied after mappings.
You can also use the macro local_ip_port to replace the properties by the local IP address and
port of the listening network of the SIP gateway used to send the INVITE.
8.
Define whether or not to enable the 180 with SDP allowed feature in the Allow 180 SDP drop-down
menu.
Table 390: 180 with SDP Parameters
Parameter
Enable
466
Description
The unit can send a SDP in the provisional response 180. Thus when the ISDN
peer sends an alerting with indication to open the voice (or if the voice is already
opened), the unit sends a 180 with SDP. This is the default value.
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Signalling Properties
Software Configuration Guide
Table 390: 180 with SDP Parameters (Continued)
Parameter
Disable
Description
A SIP 183 with SDP is sent instead of a 180 with SDP. This does not affect the 180
without SDP. This is useful if your proxy has issues receiving 180 with SDP
messages.
The SIP 183 with SDP replacing the SIP 180 with SDP is not sent if a 183 with
SDP has already been sent.
9.
Define whether or not to enable the 183 without SDP allowed feature in the Allow 183 No SDP dropdown menu.
Table 391: 183 without SDP Parameters
Parameter
10.
Description
Enable
When enabled, the unit sends a 183 without SDP upon receiving an ISDN
progress indicator without any indication to open a voice stream. This is the default
value.
Disable
When disabled, nothing is sent instead of a 183 without SDP. This does not affect
the 183 with SDP. This is useful if your proxy has issues receiving 183 without
SDP messages.
Set the privacy level of the call in the Privacy drop-down menu.
Table 392: Privacy Levels
Level
Description
Effects on incoming SIP call
Effects on outgoing SIP call
Disable No privacy
is used.
None
None
None
None
Adds two headers:
Use PAsserted
Identity
privacy.
•
Privacy: none
•
P-Asserted-Identity:
p_asserted_identity_v
alue
p_asserted_identity_value is the
call's From URI unless a SIP
header translation has been added
to the Signaling Properties for the
Identity-header.
Id
Use PPreferred
Identity
privacy.
The calling-name is empty and the
PI is set to restricted.
Always adds one header:
•
P-Preferred-Identity:
p_preferred_identity_v
alue
p_ preferred _identity_value is the
call's From URI unless a SIP
header translation has been added
to the Signaling Properties for the
Identity-header.
If the incoming call's PI property is
restricted, another header is
added:
•
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Privacy: id
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Chapter 44 - Call Router Configuration
Signalling Properties
Table 392: Privacy Levels (Continued)
Level
Rpid
Description
Use
RemoteParty-ID
privacy.
Effects on incoming SIP call
None
Effects on outgoing SIP call
One header always added :
•
Remote-Party-ID:
remote_party_id_value
"Optional Friendly
Name"<sip:410202@10.4.125.12>
;party=calling
Where remote_party_id_value
should be set by the SIP Headers
Translation. It consists of an
optional friendly name followed by
the SIP URI and the party
direction.
Example:
Remote-Party-ID: "John
Doe"<sip:410202@10.4.125.
12>;party=calling
11.
Enter the name of one or more SIP headers translation to apply to the call in the SIP Headers
Translations field.
You must define SIP headers translations as defined in “SIP Headers Translations” on page 469.
You can use the Suggestion column’s drop-down menu to select between existing translations, if
any.
You can enter more than one translation. In that case, the translations are separated with “,” and
are executed in sequential order.
12.
Enter the name of one or more call properties translation to apply to the call in the Call Properties
Translations field.
You must set call properties translations as defined in “Call Properties Translations” on page 472.
You can use the Suggestion column’s drop-down menu to select between existing translations, if
any.
You can enter more than one translation. In that case, the translations are separated with “,” and
are executed in sequential order.
13.
Click the Save button.
This brings you back to the main Call Router > Route Config web page.
You can see a yellow Yes in the Config Modified section at the top of the window. It warns you that
the configuration has been modified but not applied (i.e., the Call Router > Status differs from the
Call Router > Route Config). The Route Config sub-menu is a working area where you build up a
Call Router configuration. While you work in this area, the configured parameters are saved but not
applied (i.e., they are not used to process incoming calls). The yellow Yes flag warns you that the
configuration has been modified but is not applied.
14.
Click the Save button to enable the signalling property entry.
The current properties applied are displayed in the Call Router > Status web page. You can also
see that the yellow Config Modified Yes flag is cleared.
Examples
The following are some examples of signalling properties:
468
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Software Configuration Guide
Figure 214: Signalling Properties Examples
Moving a Signalling Property Row
The signalling properties entries sequence is very important. The call router follows the signalling properties
table rows as they are entered in the web interface. If you want the call router to try to match one row before
another one, you must put that row first.
 To move a signalling property entry up or down:
1.
Either click
or
of the row you want to move until the entry is properly located.
2.
Click the Save button to update the Call Router > Status web page.
Deleting a Signalling Property Row
You can delete a signalling property row from the table in the web interface.
 To delete a signalling property entry:
1.
Click
of the row you want to delete.
2.
Click Save to update the Call Router > Status web page.
SIP Headers Translations
A SIP Headers Translation overrides the default value of SIP headers in an outgoing SIP message. It modifies
the SIP headers before the call is sent to its destination.
Like the routing table, the SIP headers translation table finds the first matching entry. It then executes it by
modifying the behaviour of the call.
Note: You can revert back to the configuration displayed in the Call Router > Status web page at any time
by clicking the Rollback button at the bottom of the page. All modified settings in the Call Router > Route
Config page will be lost.
You can add up to 100 SIP Headers Translations.
Creating/Editing a SIP Headers Translation
The web interface allows you to create a SIP header translation or modify the parameters of an existing one.
The SIP headers translations are called from a signalling property as described in “Signalling Properties” on
page 465.
 To create or edit a SIP headers translation:
1.
In the web interface, click the Call Router link, then the Route Config sub-link.
Figure 215: Call Router – Route Config Web Page
2
3
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Chapter 44 - Call Router Configuration
SIP Headers Translations
2.
Locate the SIP Headers Translations section.
3.
Do one of the following:
•
If you want to add a SIP headers translation before an existing entry, locate the proper
row in the table and click
of this row.
•
If you want to add a SIP headers translation at the end of existing rows, click
bottom right of the SIP Headers Translations section.
•
If you want to edit an existing entry, locate the proper row in the table and click
at the
.
This brings you to the Configure SIP Headers Translation panel.
Figure 216: Configure SIP Headers Translation Panel
4
6
5
7
4.
Enter the name of the SIP headers translation in the Name field.
5.
Set which SIP header is modified by this translation in the SIP Header drop-down menu.
Table 393: SIP Headers
SIP Header
6.
Description
From Header (Host Part)
Host part of the From header's URI.
From Header (User Part)
User part of the From header's URI.
Identity Header (Host Part)
Host part of the Identity header's URI.
Identity Header (User Part)
User part of the Identity header's URI.
Identity Header (Phone Number)
Phone number in the Identity header's tel URL.
Request Line (Host Part)
Host part of the Request line's URI.
Request Line (User Part)
User part of the Request line's URI.
To Header (Host Part)
Host part of the To header's URI.
To Header (User Part)
User part of the To header's URI.
Set what information is used to build the selected SIP header in the Built From drop-down menu.
Table 394: Built From Information
Built From
470
Description
Called E164
Use the called party E.164 property.
Destination Host
Use the destination host configured in the signalling
properties of which this translation is part.
Domain
Use the domain name configured in the unit.
Fix Value
Use a fix value as defined in the Fix Value field (see Step 7).
Host Name
Use the host name configured in the unit.
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SIP Headers Translations
Software Configuration Guide
Table 394: Built From Information (Continued)
Built From
7.
Description
Local Ip
Use the local IP address.
Calling Bearer Channel
Use the calling bearer channel.
SIP Endpoint Username
Use the SIP username associated with the endpoint.
Calling Name
Use the calling party name property.
Calling E164
Use the calling party E.164 property.
If you have selected Fix Value in the Built From drop-down menu, enter a fix value to be inserted
in the SIP header in the Fix Value field.
For instance, you could hide the caller’s name in a SIP message by using the From Header (User
Part) SIP header and entering “anonymous” in the Fix Value field.
8.
Click the Save button.
This brings you back to the main Call Router > Route Config web page.
You can see a yellow Yes in the Config Modified section at the top of the window. It warns you that
the configuration has been modified but not applied (i.e., the Call Router > Status differs from the
Call Router > Route Config). The Route Config sub-menu is a working area where you build up a
Call Router configuration. While you work in this area, the configured parameters are saved but not
applied (i.e., they are not used to process incoming calls). The yellow Yes flag warns you that the
configuration has been modified but is not applied.
9.
Click the Save button to enable the SIP headers translation entry.
The current properties applied are displayed in the Call Router > Status web page. You can also
see that the yellow Config Modified Yes flag is cleared.
Example
The following is an example of SIP headers translations:
Figure 217: SIP Headers Translations Example
Moving a SIP Headers Translation Row
The SIP headers translation entries sequence is very important. The signalling properties table follows the SIP
headers translation table rows as they are entered in the web interface. If you want the signalling properties
table to try to match one row before another one, you must put that row first.
 To move a SIP headers translation entry up or down:
1.
Either click
or
of the row you want to move until the entry is properly located.
2.
Click Save to update the Call Router > Status web page
Deleting a SIP Headers Translation Row
You can delete a SIP headers translation row from the table in the web interface.
 To delete a SIP headers translation entry:
Dgw v2.0 Application
1.
Click
of the row you want to delete.
2.
Click Save to update the Call Router > Status web page.
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Chapter 44 - Call Router Configuration
Call Properties Translations
Call Properties Translations
A Call Properties Translation overrides the default value of call properties in an incoming SIP message. It
modifies the call properties before the call is sent to its destination.
Like the routing table, the call properties translation table finds the first matching entry. It then executes it by
modifying the behaviour of the call.
Note: You can revert back to the configuration displayed in the Call Router > Status web page at any time
by clicking the Rollback button at the bottom of the page. All modified settings in the Call Router > Route
Config page will be lost.
You can add up to 100 Call Properties Translations.
Creating/Editing a Call Properties Translation
The web interface allows you to create a call properties translation or modify the parameters of an existing
one. The call properties translations are called from a signalling property as described in “Signalling
Properties” on page 465.
 To create or edit a call properties translation:
1.
In the web interface, click the Call Router link, then the Route Config sub-link.
Figure 218: Call Router – Route Config Web Page
2
3
2.
Locate the Call Properties Translations section.
3.
Do one of the following:
•
If you want to add a call properties translation before an existing entry, locate the
proper row in the table and click
of this row.
•
If you want to add a call properties translation at the end of existing rows, click
the bottom right of the Call Properties Translations section.
at
•
If you want to edit an existing entry, locate the proper row in the table and click
.
This brings you to the Configure Call Properties Translation panel.
Figure 219: Configure Call Properties Translation Panel
4
6
5
7
4.
472
Enter the name of the call properties translation in the Name field.
Dgw v2.0 Application
Call Properties Translations
Software Configuration Guide
5.
Set which call property is modified by this translation in the Call Property drop-down menu.
Table 395: Call Properties
Call Property
Description
Called E164
Called party E.164 property.
Calling E164
Calling party E.164 property.
Called Name
Called party name property.
Calling Name
Calling party name property.
Called Uri
Called URI name property.
Calling Uri
Calling URI name property.
Called Bearer Channel Called bearer channel property.
6.
Set what information is used to build the selected call property in the Built From drop-down menu.
Table 396: Built From Information
Built From
Description
Domain
Use the domain name configured in the unit.
Fix Value
Use a fix value as defined in the Fix Value field (see Step 7).
From Header (Uri)
Use the From header's URI.
From Header (Friendly Name)
Use the friendly name part of the From header.
From Header (User Part)
Use the user part of the From header's URI.
Identity Header (Uri)
Use the Identity header's URI.
Identity Header (User Part)
Use the user part of the Identity header's URI.
Identity Header (Phone Number)
Use the phone number in the Identity header's tel URL. The
phone number is not retrieved if the received tel URL is
invalid. Only the phone number part is retrieved. Examples:
•
Received header: P-Preferred-Identity:
<tel:8298749;phone-context=819>
Retrieved phone number: 8298749
•
Received header: P-Preferred-Identity:
<tel:+8298749>
Retrieved phone number: 8298749
•
Received header: P-Preferred-Identity:
<tel:8298749>
Retrieved phone number: None, the received header
is invalid.
Dgw v2.0 Application
Identity Header (Friendly Name)
Use the friendly name in the Identity header's URI.
Local Ip
Use the local IP address.
Request Line (Uri)
Use the Request line's URI.
Request Line (User Part)
Use the user part of the Request line's URI.
To Header (Uri)
Use the To header's URI.
To Header (Friendly Name)
Use the friendly name part of the To header.
To Header (User Part)
Use the user part of the To header's URI.
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Chapter 44 - Call Router Configuration
7.
Call Properties Translations
If you have selected Fix Value in the Built From drop-down menu, enter a fix value to be inserted
in the call property in the Fix Value field.
For instance, you could hide the callee’s name in a SIP message by using the From Header (User
Part) SIP header and entering “anonymous” in the Fix Value field.
8.
Click the Save button.
This brings you back to the main Call Router > Route Config web page.
You can see a yellow Yes in the Config Modified section at the top of the window. It warns you that
the configuration has been modified but not applied (i.e., the Call Router > Status differs from the
Call Router > Route Config). The Route Config sub-menu is a working area where you build up a
Call Router configuration. While you work in this area, the configured parameters are saved but not
applied (i.e., they are not used to process incoming calls). The yellow Yes flag warns you that the
configuration has been modified but is not applied.
9.
Click the Save button to enable the call properties translation entry.
The current properties applied are displayed in the Call Router > Status web page. You can also
see that the yellow Config Modified Yes flag is cleared.
Example
The following is an example of call properties translations:
Figure 220: Call Properties Translations Example
Moving a Call Properties Translation Row
The call properties translation entries sequence is very important. The signalling properties table follows the
call properties translation table rows as they are entered in the web interface. If you want the signalling
properties table to try to match one row before another one, you must put that row first.
 To move a call properties translation entry up or down:
1.
Either click
or
of the row you want to move until the entry is properly located.
2.
Click Save to update the Call Router > Status web page.
Deleting a Call Properties Translation Row
You can delete a call properties translation row from the table in the web interface.
 To delete a SIP headers translation entry:
474
1.
Click
of the row you want to delete.
2.
Click Save to update the Call Router > Status web page.
Dgw v2.0 Application
Hunt Serv