Hi, thanks for the prompt reply! I have set canreinvite=no and now it works as intended. However, we want this to work with public asterisk servers, too... so is there a way to let PJSIP decline the reinvite, or otherwise enforce the RTP communication to go through the server? Thanks, Florian On 2/8/2012 06:20, J?rg Schwarzenberg wrote: > Hi! > >> we want to always use the Asterisk server as an RTP relay. How do we >> need to set up the account and/or the call for that? > > I think that is a setting on the asterisk-side. You should have a look > those: > > http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite > http://www.voip-info.org/wiki/view/Asterisk+sip+directrtpsetup > > Best Regards > J?rg Schwarzenberg > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >
