Try to run rtpproxy on private ip not on local 127.0.0.1 On Tue, Jun 28, 2011 at 11:23 AM, MingHon <gming...@gmail.com> wrote:
> 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@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- Regards, Chandrakant Solanki
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