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 -
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Book: SIP Routing With Kamailio -
http://www.asipto.com