Hi, Did you forget to fix contact and sdp in your onreply_route? Daniel
On Jun 5, 2015 3:30 AM, Alex <neiroma...@gmail.com> wrote: > > Hello! > > Please help to fix problem with sdp headers > > UAC Inet -> (X.X.X.X) Kamailio (192.168.30.250) -> Asterisk (192.168.30.2) > > When i call from UAC to 9002 i received INVITE/SDP from kamailio > > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 192.168.1.52:27080;received=10.10.101.50;branch=z9hG4bK-d8754z-027c786dac17bf68-1---d8754z-;rport=27080 > Record-Route: > <sip:192.168.30.2;line=sr-mYtaP6eErk-dx6VfrLzfr6BaPGj0OHFfPYd0OHFfPYIQpHmFr9mQPKDEx9VlvZ8QO4ttma**> > Record-Route: <sip:X.X.X.X;r2=on;lr=on;ftag=0748d948;nat=yes> > From: <sip:user4@X.X.X.X>;tag=0748d948 > To: <sip:9002@X.X.X.X>;tag=as3914e1d1 > Call-ID: ZWU5YmFiNTNhNmNmYWQzYzhkZWUzZDNjOTU3MDFiNGU. > CSeq: 2 INVITE > Server: Virtel.net Node2 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH, MESSAGE > Supported: replaces, timer > Contact: <sip:192.168.30.2;line=sr-mYtaP62ar9nzrg20y6eYPA-LrA-0P6Bax6z*> > Content-Type: application/sdp > Content-Length: 278 > > v=0 > o=root 732368067 732368067 IN IP4 X.X.X.X > s=Asterisk PBX 11.17.1 > c=IN IP4 X.X.X.X > t=0 0 > m=audio 15768 RTP/AVP 0 8 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=sendrecv > a=nortpproxy:yes > > Why Record-Route and Contact fields contain private IP of asterisk ? > _______________________________________________ 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
