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 ?
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