[asterisk-users] Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)

[Michael Ulitskiy]

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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