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

Reply via email to