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 .
Hi Ryan,
if you didnt use the nating Route - rtpproxy_manage() would never called and so rtpproxy didnt work.
Try to use rtpproxy_manage and use xlog to show that is fired up.
2012/4/11 Ryan Gholam ryangholam@gmail.com:
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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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Karsten Horsmann -
Ryan Gholam