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 ?