You should post a SIP trace, together with the IP addresses of all nodes: ngrep -t -d any -P "" -Wbyline port 5060
If there is sensitive information in the traces, just remove/replace it. regards Klaus Am 21.07.2011 09:23, schrieb MingHon: > Hello List, > > im still trying but no luck. > asterisk canreinvite already set to yes > > now im testing in lan > i setup kamailio and asterisk in same lan > kamailio&rtpproxy on 192.168.2.3 and asterisk on 192.168.2.23 > > canreinvite=yes in asterisk. when both ua in the same lan > register directly to asterisk the reinvite work. both ua will have > and direct media flow > > [ua1]<====>[ua2] > | > | > x > | > v > [asterisk] > > when ua register to kamailio the audio work and the reinvite message is > same as the first invite message. > > [ua1]<====>[kamailio]<====>[ua2] > | ^ > | | > | | > v | > [asterisk] > > how do i stop the media flow between kamailio and asterisk? > make kamailio relay the rtp between both ua. > > [ua1]<====>[kamailio]<====>[ua2] > | ^ > x x > | | > v | > [asterisk] > > > anyone could give some hint? > > thanks in adv. > > -- > 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 _______________________________________________ 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
