Using Asterisk 18.12.0, a little confused on how to configure my pjsip.conf file to determine the codec to use for a call I have 2 endpoints:[Alice]disallow:all allow:ulaw,alaw,g729[Bob]disallow:all allow:ulaw,alaw,g729Alice calls into Asterisk on ..
Author : Fabian Borot
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Asterisk receives a 180 Ringing with SDP from the called side, then it sends 180 without SDP to the calling side.We would like asterisk to sends to the calling side the same response that was received from the called side. This is Asterisk cert 13..
In Version 1.8 asterisk introduced this parameter preferred_codec_only, when set to yes the 200 OK to the INVITE contains 1 codec only from the available ones in the user sip profile.But in version 13.1 (I think version 11.2 also) is not working l..