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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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