Hi, have you tried "canreinvite=yes" on your Asterisk-box? If that does not help, there is probably no way to make the RTP-Traffic bypass your asterisk box...
Carsten 2011/7/14 MingHon <gming...@gmail.com>: > Hello, > anyone? > currently my setup look like this. > when UA1 call UA2 and the rtp traffic flow to kamailio and asterisk. > [UA1] <--(rtp)--> [Kamailio/RTPProxy] <--(rtp)--> [UA2] > ^ > | > RTP TRAFFIC > | > v > [ASTERISK] > what i need to achieve is UA1 call UA2 and rtp traffic flow to kamailio but > not to asterisk. > can kamailio handle the rtp traffic it own? > [UA1] <--(rtp)--> [Kamailio/RTPProxy] <--(rtp)--> [UA2] > ^ > | > X > | > v > [ASTERISK] > > Thanks in advance. > -- > Regards, > > MingHon > -- Carsten Bock http://www.ng-voice.com mailto:cars...@ng-voice.com Schomburgstr. 80 22767 Hamburg Germany Mobile +49 179 2021244 Office +49 40 34927219 Fax +49 40 34927220 ~~~~~ Checkout SIP-Provider CE: http://www.sipwise.com/products/spce/overview/ _______________________________________________ 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
