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 ?