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<p>Hello,</p>
<p>Anyone? I have hard time to believe this is not possible with
chan_pjsip.</p>
<p>Anyway, may I ask how people handle the following scenario which
I imagine should be quite common:</p>
- I have internal extensions talk to each other using g722. so their
codec setting (with chan_sip now) is "allow=g722,ulaw"<br>
- I have carriers trunks that handle ulaw only (allow=ulaw)<br>
- calls between internal extensions naturally happen over g722 as
its their preferred codec<br>
- for external calls I now set SIP_CODEC_INBOUND=ulaw to influence
codec selection on calling channel and the calls set up using ulaw
end-to-end
<p>Can somebody please advise how to achieve the same with
chan_pjsip?</p>
<p>Thanks,</p>
<p>Michael<br>
</p>
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<div class="moz-cite-prefix">On 6/30/23 09:30, Michael Ulitskiy
wrote:<br>
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cite="mid:167fd103-8f55-4403-ad5a-9eddd35359e8@acedsl.com">
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<p>Hello,<br>
</p>
<p>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.<br>
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 <em>calling</em>
channel codec selection from dialplan?<br>
I’m working with asterisk 20.3.0.<br>
</p>
<p>Thank you,<br>
Michael</p>
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