Am 13.07.2011 10:07, schrieb MingHon: > Hi, > > i tried set canreinvite=yes in asterisk but kamailio/rtpproxy still > trying to send rtp traffic to asterisk.
That should not happen. You have to investigate why. You have to take a look at the SIP signaling during and after call setup. You should see reINVITE messages from Asterisk to the clients. Take a look at the SDPs in those requests and their responses to find out if they are malformed. regards Klaus _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
