FYI: i've created a feature request to add SIP_CODEC_INBOUND equivalent
functionality to chan_pjsip:
https://github.com/asterisk/asterisk-feature-requests/issues/9
Let's see where it goes
*Michael Ulitskiy*
Ace Innovative Networks, Inc.
Main/SMS: 212-868-2366
Direct/SMS: 212-812-1203
https://www.aceinnovative.com
On 7/5/23 11:58, Michael Ulitskiy wrote:
Hello,
Anyone? I have hard time to believe this is not possible with chan_pjsip.
Anyway, may I ask how people handle the following scenario which I
imagine should be quite common:
- I have internal extensions talk to each other using g722. so their
codec setting (with chan_sip now) is "allow=g722,ulaw"
- I have carriers trunks that handle ulaw only (allow=ulaw)
- calls between internal extensions naturally happen over g722 as its
their preferred codec
- 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
Can somebody please advise how to achieve the same with chan_pjsip?
Thanks,
Michael
On 6/30/23 09:30, Michael Ulitskiy wrote:
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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