On Thu, Aug 15, 2019, at 8:23 AM, Jöran Vinzens wrote: > Hi All, > > We are using asterisk 16.5 and having an issue with the first re-invite > after the call has been established. > We can see the call gets up and you see in the logs the bridge type has > changed and after that a re-invite is triggered. > > Is there any possibility to deactivate this kind of reInvite? We have > some race conditions while have multiple asterisk in the call flow and > the different asterisk systems are sending this reInvites out parallel. > While an invite is pending on a system it is not accepting another > incoming reInvite from peer. > > With chan_SIP canreinvite=no solved the issue. But it seems there is > nothing similar in PJSIP.
The "direct_media" option controls just that, direct media. A reinvite can occur for other reasons (such as attempting to renegotiate streams to be of better quality or to update connected line information). Have you determined which case is occurring? -- 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
