Hello,
I'm using asterisk-unimrcp via the exposed dialplan applications (app_unimrcp.so) rather than through the Generic Speech API (res_speech_unimrcp.so).
I wanted to ask if it's possible to specify the "sip-t1x64" timeout on SIP INVITEs that do not get replies, under app_unimrcp.so?
My understanding is that the answer is probably no due to the following, but wanted to double-check...
- The exposed SIP Timeouts (sip-t1, sip-t2, sip-t4 and sip-t1x64) can be specified in unimrcpclient.xml or one of its referenced XML files.
- Generic Speech API (res_speech_unimrcp.so) appears to be the sole module that utilizes unimrcpclient.xml.
- app_unimrcp.so appears to only use mrcp.conf.
I am successfully using the request-timeout property to limit the RTSP timeout via mrcp.conf, but curious if similar is possible on the SIP level.
Thanks very much.
Sincerely,
Stephen