Re: [SR-Users] Kamailio Failed to have audo converstaion with RTP

Dear Daniel , I thank you for your reply , I have a server  having the Astersisk  ip address (192.168.10.15) , and rtp + kamailio is installed on an another pc have the  following ip (192.168.10.17) which is linked to the Astersik , and on the same pc , another network card exists having the ip address (192.168.20.3 ) which is linked to a client pc having the ip address ( 192.168.20.4) . I tracked the call and i can see  SIP ACK nd BYE between 20.3 and 20.4 but there is no audio conversation   this is my configuration file for kamailio attached above . P.S :  testing without  NATING as described in the above setup  . I thank you alot again for all your help . _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users