Re: Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)
Michael Ulitskiy <[email protected]> Thu, 6 Jul 2023 13:22:19 -0400
| Newsgroups | gmane.comp.telephony.pbx.asterisk.user |
|---|---|
| Organization | Ace Innovative Networks, Inc |
| Message-ID | <[email protected]> |
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 <[email protected]> > 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> > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users