On Wed, Dec 12, 2018, at 12:31 PM, Michael Maier wrote: <snip>
> > The problem: The extension doesn't create a ringback locally, because > it most probably expects it to > be sent by the callee - but the callee doesn't send anything (not > surprising, because there has been > no SDP). > > Or should Asterisk create the ringback (Asterisk doesn't send any RTP > package)? Or should the phone > create the ringback itself because there is a 180 Ringing (even if it > contains SDP)? > > I'm wondering: Why does Asterisk create a 183 to the extension > containing SDP if the callee didn't > provide any SDP? > > > So many questions ... . Could somebody please shine some light on it? > What's going wrong here? The core doesn't communicate whether progress includes media or not, so the PJSIP channel driver (and even chan_sip) assumes media is there. What should happen in chan_pjsip is that it would send inband ringing and not a 180 Ringing with SDP, but that is not currently implemented and I don't think this particularly interaction has come up before to cause it to be implemented. I'd therefore suggest raising an issue[1] with the SIP trace. [1] https://issues.asterisk.org/jira -- Joshua C. Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & 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
