Mapping PSTN Release Cause to SIP Response for Analog Interfaces

The device's FXO interface interoperates 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 IP-to-Tel calls. The converse is also true - for Tel-to-IP calls, the SIP 4xx or 5xx responses are mapped to tones played to the PSTN/PBX.

When establishing an IP-to-Tel call, the following rules are applied:

■

If the remote party (PSTN/PBX) is busy and the FXO device detects a busy tone, it sends a SIP 486 Busy response to IP. If it detects a reorder tone, it sends a SIP 404 Not Found (no route to destination) to IP. In both cases the call is released. Note that if the 'Disconnect Call on Busy Tone Detection' parameter is set to Disable, the FXO device ignores the detection of busy and reorder tones and doesn't release the call.

■

For all other FXS/FXO release types such as:

●

no free channels in the Trunk Group,

●

an appropriate call routing rule to a Trunk Group doesn’t exist, or

●

the phone number isn’t found

the device sends a SIP response to the IP according to the 'Default Release Cause' parameter. The parameter defines Q.931 release causes. Its default value is 3, which is mapped to the SIP 404 response. By changing its value to 34, the SIP 503 response is sent. Other causes can be used as well.