Advanced Pstn Configuration; Release Reason Mapping - AudioCodes Mediant 1000 User Manual

Voip media gateways, sip protocol
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Mid-call communication: After the SIP connection is established, all QSIG
messages are encapsulated in SIP INFO messages.
Call tear-down: The SIP connection is terminated once the QSIG call is complete.
The Release Complete message is encapsulated in the SIP BYE message that
terminates the session.
To enable QSIG tunneling:
1.
Set the EnableQSIGTunneling parameter to 1 on the originating and terminating
devices.
2.
Configure the QSIGTunnelingMode parameter for defining the format of encapsulated
QSIG message data in the SIP message MIME body (0 for ASCII presentation; 1 for
binary encoding).
3.
Set the ISDNDuplicateQ931BuffMode parameter to 128 to duplicate all messages.
4.
Set the ISDNInCallsBehavior parameter to 4096.
5.
Set the ISDNRxOverlap parameter to 0 for tunneling of QSIG overlap-dialed digits
(see below for description).
The configuration of the ISDNInCallsBehavior and ISDNRxOverlap parameters allows
tunneling of QSIG overlap-dialed digits (Tel to IP). In this configuration, the device delays
the sending of the QSIG Setup Ack message upon receipt of the QSIG Setup message.
Instead, the device sends the Setup Ack message to QSIG only when it receives the SIP
INFO message with Setup Ack encapsulated in its MIME body. The PBX sends QSIG
Information messages (to complete the Called Party Number) only after it receives the
Setup Ack. The device relays these Information messages encapsulated in SIP INFO
messages to the remote party.

18.1.6 Advanced PSTN Configuration

This section describes various advanced PSTN configurations.

18.1.6.1 Release Reason Mapping

This section describes the available mapping mechanisms of SIP responses to Q.850
Release Causes and vice versa. The existing mapping of ISDN Release Causes to SIP
Responses is described in 'Fixed Mapping of ISDN Release Reason to SIP Response' on
page
243
and 'Fixed Mapping of SIP Response to ISDN Release Reason' on page 245. To
override this hard-coded mapping and flexibly map SIP responses to ISDN Release
Causes, use the ini file (CauseMapISDN2SIP and CauseMapSIP2ISDN, as described in
'ISDN and CAS Interworking Parameters' on page 698) or the Web interface (see
'Configuring Release Cause Mapping' on page 267).
It is also possible to map the less commonly used SIP responses to a single default ISDN
Release Cause. Use the parameter DefaultCauseMapISDN2IP (described in 'ISDN and
CAS Interworking Parameters' on page 698) to define a default ISDN Cause that is always
used except when the following Release Causes are received: Normal Call Clearing (16),
User Busy (17), No User Responding (18) or No Answer from User (19). This mechanism
is only available for Tel-to-IP calls.
18.1.6.1.1 Reason Header
The device supports the Reason header according to RFC 3326. The Reason header
conveys information describing the disconnection cause of a call:
Sending Reason header: If a call is disconnected from the Tel side (ISDN), the
Reason header is set to the received Q.850 cause in the appropriate message
(BYE/CANCEL/final failure response) and sent to the SIP side. If the call is
disconnected because of a SIP reason, the Reason header is set to the appropriate
SIP response.
SIP User's Manual
242
Mediant 600 & Mediant 1000
Document #: LTRT-83310

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