Hello, On 05/06/15 21:39, Alex 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 ? as a guess based on what I can see in the pasted reply, you are using topoh module and mask_ip is set to 192.168.30.2.
For better understanding of what you do, you have to provide full sip trace, all incoming and outgoing sip messages from initial INVITE to the 200ok for INVITE sent to caller. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com
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