MP-1xx SIP User’s Manual 8. Gateway Capabilities Description
Version 4.4 157 March 2005
• Detection of silence -
The call is disconnected after silence is detected on both call directions for a specific
(configurable) amount of time. The call isn’t disconnected immediately; therefore, this
method should only be used as a backup.
Relevant ini file parameters: EnableSilenceDisconnect and FarEndDisconnectSilencePeriod
(with DSP templates number 2 or 3).
• Interruption of RTP stream -
Relevant ini file parameters: BrokenConnectionEventTimeout and
DisconnectOnBrokenConnection. Note that this method operates correctly only if silence
suppression is not used.
• Protocol-based termination of the call from the IP side.
8.6 Mapping PSTN Release Cause to SIP Response
The MP-1xx FXO gateway is used to interoperate between the SIP network and the PSTN/PBX.
This interoperability includes the mapping of PSTN/PBX Call Progress Tones to SIP 4xx or 5xx
responses for IPTel calls. The converse is also true: For TelIP calls, the SIP 4xx or 5xx
responses are mapped to tones played to the PSTN/PBX.
When establishing an IPTel call the following rules are applied:
If the remote party (PSTN/PBX) is busy and the FXO gateway detects a Busy tone, it sends 486
busy to IP. If it detects a Reorder tone, it sends 404 not found (no route to destination) to IP. In
both cases the call is released. Note that if ‘DisconnectOnBusyTone = 0’ the FXO gateway
ignores the detection of Busy/Reorder tones and doesn’t release the call.
For all other MP-1xx FXS/FXO releases (caused when there are no free channels in the specific
hunt group, or when an appropriate rule for routing the call to a hunt group doesn’t exist, or if the
phone number isn’t found), the MP-1xx sends SIP response (to IP) according to the parameter
‘DefaultReleaseCause’. This parameter defines Q.931 release causes. Its default value is ‘3’, that
is mapped to SIP 404 response. By changing its value to ‘34’ SIP 503 response is sent. Other
causes can be used as well.
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