[asterisk-users] Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)
Joshua C. Colp
jcolp at sangoma.com
Thu Jul 6 12:32:58 CDT 2023
On Thu, Jul 6, 2023 at 2:22 PM Michael Ulitskiy <mulitskiy at acedsl.com>
wrote:
> 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.
>
The wiki is read-only now and documentation has moved to
https://docs.asterisk.org/, I have updated the page[1] and it will deploy
in a few minutes.
> Thanks for clarifying it. Any idea on the timeframe of implementation?
>
There is no timeframe on such a thing.
[1]
https://docs.asterisk.org/20/Development/Roadmap/Asterisk-18-Projects/Advanced-Codec-Negotiation-ACN/?h=advanced+codec
--
Joshua C. Colp
Asterisk Project Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20230706/0f175743/attachment.html>
More information about the asterisk-users
mailing list