Re: [SR-Users] rtpproxy and kamailio doesnt work out from the box.

Hi, i fixed the audio issue for 102 to 103 vice versa. by fixing the canreinvite in asterisk. from uac the rtp packet will route to kamailio den forward to asterisk. can we bypass the rtp packet going to asterisk? and here is the update for uac 101 issue. when 101 call to voicemail or 102/103 there is no audio. in wireshark i saw 101 send rtp packet to a private ip belong to asterisk. but if 102/103 call to 101 both uac got audio. i realize this is because 101 is the first uac registered before 102/103 and because it did not have the received: field in ul show. please adv. -- Regards, MingHon _______________________________________________ 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