Hi, i tried set canreinvite=yes in asterisk but kamailio/rtpproxy still trying to send rtp traffic to asterisk.
and asterisk did not forward the rtp traffic back to kamailio/rtpproxy then i will get no audio on the ua. please adv. thanks, Regards, MingHon
_______________________________________________ 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