Hi all, So the scenario is:
A -> Asterisk -> B after B send back 200 OK Asterisk is answering the call to A. Directly after the Answer Asterisk generates a ReInvite to A and the only difference between the 200 OK sdp and the reInvite sdp are the offered codecs which are forwarded from B to A. Here i do not understand why this could not be done in the 200OK to A? As far as i understood you Josh, there is no way to prohibit this kind of reInvite? It is not about route Optimization just for some more options for the A Party. BR Jöran On Thu, Aug 15, 2019 at 4:07 PM Joshua C. Colp <[email protected]> wrote: > 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 -- Jöran Vinzens - [email protected] sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.co.uk
-- _____________________________________________________________________ -- 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
