Hello,
I finally got to look at chan_sip to chan_pjsip migration again. This
time I’m having problems with influencing codec selection on originating
(calling) channel. It looks like PJSIP_MEDIA_OFFER only works on
outbound (called) channel and has no affect on calling channel. My
experiments and function documentation (which says “Media and codec
offerings to be set on an outbound SIP channel prior to dialing.”) seem
to confirm it.
So PJSIP_MEDIA_OFFER is supposed to be (and it works) chan_pjsip’s
equivalent of ${SIP_CODEC_OUTBOUND}, but what is chan_pjsip’s equivalent
of ${SIP_CODEC_INBOUND}? Or, in other words, what are we supposed to do
to influence /calling/ channel codec selection from dialplan?
I’m working with asterisk 20.3.0.
Thank you,
Michael
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