Re: Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)
"Joshua C. Colp" <[email protected]> Thu, 6 Jul 2023 13:47:04 -0300
| Newsgroups | gmane.comp.telephony.pbx.asterisk.user |
|---|---|
| Message-ID | <CAM0A2Z1hgCu5pCJFuYmvQ2TG94BBcV4nrK9EMSu8MjMd-Jn4MA@mail.gmail.com> |
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 and 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