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Hello,<br>
<br>
<div class="moz-cite-prefix">On 05/06/15 21:39, Alex wrote:<br>
</div>
<blockquote
cite="mid:CADfB73eDb7eDJu2sA+O3X9NJF_qKNL=A_00E522X9j+SVeL-gQ@mail.gmail.com"
type="cite">
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<div style="font-size:12.8000001907349px">Hello!</div>
<div style="font-size:12.8000001907349px"><br>
</div>
<div style="font-size:12.8000001907349px">Please help to fix
problem with sdp headers</div>
<div style="font-size:12.8000001907349px"><br>
</div>
<div style="font-size:12.8000001907349px">UAC Inet ->
(X.X.X.X) Kamailio (192.168.30.250) -> Asterisk
(192.168.30.2)</div>
<div style="font-size:12.8000001907349px"><br>
</div>
<div style="font-size:12.8000001907349px">When i call from UAC
to 9002 i received INVITE/SDP from kamailio<br>
</div>
<div style="font-size:12.8000001907349px"><br>
</div>
<div style="font-size:12.8000001907349px"> <span class="">SIP</span>/2.0
200 OK</div>
<div style="font-size:12.8000001907349px"> Via: <span
class="">SIP</span>/2.0/UDP
192.168.1.52:27080;received=10.10.101.50;branch=z9hG4bK-d8754z-027c786dac17bf68-1---d8754z-;rport=27080</div>
<div style="font-size:12.8000001907349px"> Record-Route: <<span
class="">sip</span>:192.168.30.2;line=sr-mYtaP6eErk-dx6VfrLzfr6BaPGj0OHFfPYd0OHFfPYIQpHmFr9mQPKDEx9VlvZ8QO4ttma**></div>
<div style="font-size:12.8000001907349px"> Record-Route: <<span
class="">sip</span>:X.X.X.X;r2=on;lr=on;ftag=0748d948;nat=yes></div>
<div style="font-size:12.8000001907349px"> From: <<span
class="">sip</span>:user4@X.X.X.X>;tag=0748d948</div>
<div style="font-size:12.8000001907349px"> To: <<span
class="">sip</span>:9002@X.X.X.X>;tag=as3914e1d1</div>
<div style="font-size:12.8000001907349px"> Call-ID:
ZWU5YmFiNTNhNmNmYWQzYzhkZWUzZDNjOTU3MDFiNGU.</div>
<div style="font-size:12.8000001907349px"> CSeq: 2 INVITE</div>
<div style="font-size:12.8000001907349px"> Server: Virtel.net
Node2</div>
<div style="font-size:12.8000001907349px"> Allow: INVITE,
ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE</div>
<div style="font-size:12.8000001907349px"> Supported:
replaces, timer</div>
<div style="font-size:12.8000001907349px"> Contact: <<span
class="">sip</span>:192.168.30.2;line=sr-mYtaP62ar9nzrg20y6eYPA-LrA-0P6Bax6z*></div>
<div style="font-size:12.8000001907349px"> Content-Type:
application/sdp</div>
<div style="font-size:12.8000001907349px"> Content-Length:
278</div>
<div style="font-size:12.8000001907349px"> </div>
<div style="font-size:12.8000001907349px"> v=0</div>
<div style="font-size:12.8000001907349px"> o=root 732368067
732368067 IN IP4 X.X.X.X</div>
<div style="font-size:12.8000001907349px"> s=Asterisk PBX
11.17.1</div>
<div style="font-size:12.8000001907349px"> c=IN IP4 X.X.X.X</div>
<div style="font-size:12.8000001907349px"> t=0 0</div>
<div style="font-size:12.8000001907349px"> m=audio 15768
RTP/AVP 0 8 101</div>
<div style="font-size:12.8000001907349px"> a=rtpmap:0
PCMU/8000</div>
<div style="font-size:12.8000001907349px"> a=rtpmap:8
PCMA/8000</div>
<div style="font-size:12.8000001907349px"> a=rtpmap:101
telephone-event/8000</div>
<div style="font-size:12.8000001907349px"> a=fmtp:101 0-16</div>
<div style="font-size:12.8000001907349px"> a=ptime:20</div>
<div style="font-size:12.8000001907349px"> a=sendrecv</div>
<div style="font-size:12.8000001907349px"> a=nortpproxy:yes</div>
<div style="font-size:12.8000001907349px"><br>
</div>
<div style="font-size:12.8000001907349px">Why Record-Route and
Contact fields contain private IP of asterisk ?</div>
</div>
</blockquote>
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.<br>
<br>
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.<br>
<br>
Cheers,<br>
Daniel<br>
<pre class="moz-signature" cols="72">--
Daniel-Constantin Mierla
<a class="moz-txt-link-freetext" href="http://twitter.com/#!/miconda">http://twitter.com/#!/miconda</a> - <a class="moz-txt-link-freetext" href="http://www.linkedin.com/in/miconda">http://www.linkedin.com/in/miconda</a>
Book: SIP Routing With Kamailio - <a class="moz-txt-link-freetext" href="http://www.asipto.com">http://www.asipto.com</a></pre>
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