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Version 5.6 111 November 2008
SIP User's Manual 3. Web-Based Management
Parameter Description
[1] Enable.
Note: P-Associated-URIs in registration responses is handled only if the
device is registered per endpoint.
Source Number
Preference
[SourceNumberPreferen
ce]
Determines the SIP header used to determine the Source Number in
incoming INVITE messages.
“” (empty string) = Use device's internal logic for header preference
(default).
“FROM” = Use the Source Number received in the From header.
The valid range is a string of up to 10 characters. The default is an
empty string.
Forking Handling Mode
[ForkingHandlingMode]
Determines how the device reacts to forking of outgoing INVITE
messages by the Proxy.
[0] Sequential handling = The device opens a voice stream toward
the first 18x SIP response that includes an SDP, and disregards any
18x response with an SDP received thereafter (default).
[1] Parallel handling = The device opens a voice stream toward the
first 18x SIP response that includes an SDP, and re-opens the
stream toward any subsequent 18x responses with an SDP.
Note: Regardless of the ForkingHandlingMode value, once a SIP 200
OK response is received, the device uses the RTP information and re-
opens the voice stream, if necessary.
Enable Reason Header
[EnableReasonHeader]
Enables / disables the usage of the SIP Reason header.
[0] Disable.
[1] Enable (default).
Retransmission Parameters
SIP T1 Retransmission
Timer [msec]
[SipT1Rtx]
The time interval (in msec) between the first transmission of a SIP
message and the first retransmission of the same message.
The default is 500.
Note: The time interval between subsequent retransmissions of the
same SIP message starts with SipT1Rtx and is multiplied by two until
SipT2Rtx. For example (assuming that SipT1Rtx = 500 and SipT2Rtx =
4000):
The first retransmission is sent after 500 msec.
The second retransmission is sent after 1000 (2*500) msec.
The third retransmission is sent after 2000 (2*1000) msec.
The fourth retransmission and subsequent retransmissions until
SIPMaxRtx are sent after 4000 (2*2000) msec.
SIP T2 Retransmission
Timer [msec]
[SipT2Rtx]
The maximum interval (in msec) between retransmissions of SIP
messages.
The default is 4000.
Note: The time interval between subsequent retransmissions of the
same SIP message starts with SipT1Rtx and is multiplied by two until
SipT2Rtx.
SIP Maximum RTX
[SIPMaxRtx]
Maximum number of UDP transmissions (first transmission plus
retransmissions) of SIP messages.
The range is 1 to 30. The default value is 7.
- User's Manual 1
- Table of Contents 3
- List of Figures 8
- List of Tables 11
- WEEE EU Directive 13
- Customer Support 13
- Trademarks 13
- Related Documentation 14
- 1 Overview 15
- 1.2 MediaPack Features 16
- 1.3 SIP Overview 17
- Reader’s Notes 18
- 2 Configuration Concepts 19
- 3 Web-Based Management 21
- 3.3.1 Toolbar 24
- 3.3.2 Navigation Tree 25
- 3.3.3.1 Accessing Pages 28
- 3.3.3.2 Viewing Parameters 28
- 3.3.3.5 Working with Tables 32
- 3.3.5.1 Creating a Scenario 36
- 3.3.5.3 Editing a Scenario 39
- 3.3.5.6 Deleting a Scenario 42
- 3.3.7 Getting Help 47
- 3.3.8 Using the Home Page 48
- (grey): channel inactive 49
- 3.4 Configuration Tab 52
- [0] = OAMP (default) 65
- [1] = Media 65
- [2] = Control 65
- 3.4.2 Media Settings 67
- [2] Bypass 70
- [3] Events Only 70
- 3.4.3 Security Settings 79
- [1] Enable = Enabled 116
- [0] Disable (default) 116
- [1] Enable 116
- 5 steps: 120
- To the default Proxy 121
- 3.4.4.1.4 Coders 123
- 4 steps: 130
- 3.4.4.2.3 Metering Tones 144
- 3.4.4.2.5 Keypad Features 147
- 3.4.4.6.1 Authentication 175
- 3.4.4.6.2 Automatic Dialing 176
- 3.4.4.6.4 Call Forward 178
- 3.4.4.6.6 Call Waiting 180
- Disable (default) 188
- No = Don't register 189
- Yes = Register 189
- 3.4.5 Advanced Applications 190
- [1] Enable (Bellcore) 194
- [2] Ericsson MD-110 194
- [3] NEC (ICS) 194
- 3.5 Management Tab 198
- 3.5.1.3 Maintenance Actions 207
- 2 steps: 209
- 3.5.2 Software Update 210
- 3.6.2 Gateway Statistics 223
- 3.6.2.1 Call Counters 224
- 3.6.2.2 Call Routing Status 226
- 3.6.2.3 Registration Status 227
- 3.6.2.5 IP Connectivity 228
- 4 ini File Configuration 231
- 4.2 The ini File Structure 232
- 4.3 Modifying an ini File 235
- 4.4.1 Networking Parameters 236
- 4.4.2 System Parameters 244
- [0] = Disabled (default) 251
- 4.4.4 Security Parameters 252
- 4.4.5 RADIUS Parameters 257
- 4.4.6 SNMP Parameters 258
- [1] = Enabled 268
- Port = Port number 268
- UserId = User name 272
- UserPassword = Password 272
- 4.4.8 Voice Mail Parameters 277
- 4.4.9 PSTN Parameters 279
- [66] = TBR21 280
- Npi = Number Plan 297
- Ton = Type of Number 297
- 4.4.12 Channel Parameters 298
- [0] = Disable (default) 299
- [1] = Enable 299
- [1] Enable (default) 304
- 5 Default Settings 305
- 6.4 Coefficient File 312
- 6.5 User Information File 313
- 7 IP Telephony Capabilities 315
- 7.1.1 Configuring SAS 316
- 7.3.2.1 T.38 Fax Relay Mode 319
- 7.3.3 Supporting V.34 Faxes 324
- 7.4 FXO Operating Modes 326
- 7.4.1.1 One-Stage Dialing 327
- 7.4.1.2 Two-Stage Dialing 328
- 7.4.1.4 DID Wink 329
- 7.4.2 Telephone-to-IP Calls 330
- Connectivity and QoS) 335
- 7.11 Call Detail Record 339
- 7.13 Configuration Examples 342
- 7.14.3 Call Transfer 359
- 7.14.4 Call Forward 360
- 7.14.5 Call Waiting 361
- 7.14.7 Caller ID 362
- 8 Networking Capabilities 365
- 8.2.1 STUN 366
- 8.2.3 No-Op Packets 367
- 8.3 IP Multicasting 368
- 8.8 VLANS and Multiple IPs 370
- Control: Premium control 372
- Management: Bronze 372
- 10.1 MP-11x Specifications 381
- 10.2 MP-124 Specifications 384
- 11 Glossary 387
Verwandte Produkte und Handbücher für Gateways / Controller AudioCodes MediaPack 114
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