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

[Michael Ulitskiy]

Oh, that's great. It wasn't clear from that page, at least not for me. :-(

Having it clearly stated on the document would save me (and probably 
others) lots of time.

Thanks for clarifying it. Any idea on the timeframe of implementation?

*Michael Ulitskiy*
Ace Innovative Networks, Inc.
Main/SMS: 212-868-2366
Direct/SMS: 212-812-1203
https://www.aceinnovative.com
On 7/6/23 12:47, Joshua C. Colp wrote:
> On Thu, Jul 6, 2023 at 1:43 PM Michael Ulitskiy <mulitskiy at acedsl.com> 
> wrote:
>
>     Hello,
>
>     After I have re-read the "PJSIP Advanced Codec negotiation"
>     document, it occurred to me that the desired behavior should
>     actually happen automatically, just due to the codec negotiation
>     logic, but it looks like asterisk doesn't actually follow the
>     described logic which is likely a bug.
>
>
> That functionality is not implemented as of this time.
>
> -- 
> Joshua C. Colp
> Asterisk Project Lead
> Sangoma Technologies
> Check us out at www.sangoma.com <http://www.sangoma.com> and 
> www.asterisk.org <http://www.asterisk.org>
>
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