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 like that , it keeps sending all the codecs and sometimes both parties pick a different one causing one way audio. Example: INVITE has ulaw, alaw, gsm and 200 OK from asterisk has alaw, g729,ulaw. Then a media capture shows the calling side sending ulaw and the asterisk sends alaw causing one way audio. Is this happening to anybody else?
This is the description of the parameter from the sip.conf