Hi, If you will send call without answering on asterisk and have directrtpsetup=yes in sip.conf codec negociation will always be between UAs so any matched codec will work fine. If you are answering call on asterisk then dialing it out to next UA then you need to add canreinvite=yes for both UAs.
Regards, Faisal P peers calling each other: A (g722, alaw) calls B (alaw,ulaw) via asterisk. My setup: allow=g722,alaw preferred_codec_only=no Note that when B calls A, codec alaw is used on both ends, fine, but it does not seem to work the reverse way (A is using g722, B is using alaw, asterisk is doing transcoding). Is it possible? Thanks, Ondrej
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