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    Hello,<br>
    <br>
    can you show both received 200ok + ACK as well as those sent out? It
    is important to see how Record-/Route, Contact and r-uri change on
    the way to spot where the issue is.<br>
    <br>
    Cheers,<br>
    Daniel<br>
    <br>
    <div class="moz-cite-prefix">On 12/05/15 05:56, Darren Campbell
      (Primar) wrote:<br>
    </div>
    <blockquote
      cite="mid:66CCCC287FD4D646A255F1495F204AB528646A0D@MBX-02.mtrx.com.au"
      type="cite">
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      <div style="direction: ltr;font-family: Tahoma;color:
        #000000;font-size: 10pt;">Hi all<br>
        <br>
        Experiencing a commonly reported issue where calls drop out
        after 30 seconds or so. Mainly because the provider hangs up
        after not recognising/receiving ACK in response to 200 OK.<br>
        <br>
        Unfortunately (or maybe fortunately), I haven't had much
        experience with Enswitch so was hoping someone in the community
        might help guide me as to which rules Enswitch might be using to
        match ACKs to calls in progress. Maybe there is another avenue I
        should be investigating.<br>
        <br>
        <br>
        Here's a sample of the 200 OK and ACK that repeats.<br>
        <br>
        13:44:04.155646 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP,
        length: 1058<br>
        E..>.M..?..Ug.v..........*J.SIP/2.0 200 OK^M<br>
        Via: SIP/2.0/UDP
172.21.0.226;rport=5060;branch=z9hG4bKfe94.efbf7fbcaf8bd15243a61fdc9d6d1e78.0^M<br>
        Via: SIP/2.0/UDP
        127.0.0.1:5080;branch=z9hG4bK65f00a0c;rport=5080^M<br>
        Record-Route: <a class="moz-txt-link-rfc2396E" href="sip:PROVIDERIP;lr=on"><sip:PROVIDERIP;lr=on></a>^M<br>
        Record-Route:
        <a class="moz-txt-link-rfc2396E" href="sip:172.21.0.226;r2=on;lr=on;ftag=as65919d92;nat=yes"><sip:172.21.0.226;r2=on;lr=on;ftag=as65919d92;nat=yes></a>^M<br>
        Record-Route:
        <a class="moz-txt-link-rfc2396E" href="sip:127.0.0.1;r2=on;lr=on;ftag=as65919d92;nat=yes"><sip:127.0.0.1;r2=on;lr=on;ftag=as65919d92;nat=yes></a>^M<br>
        From: "asterisk"
        <a class="moz-txt-link-rfc2396E" href="sip:PROVIDERUSER@PROVIDERIP:5080"><sip:PROVIDERUSER@PROVIDERIP:5080></a>;tag=as65919d92^M<br>
        To: <a class="moz-txt-link-rfc2396E" href="sip:PHONENUMBER@PROVIDERIP"><sip:PHONENUMBER@PROVIDERIP></a>;tag=as260fefaa^M<br>
        Call-ID: 271ac7a174d613cd0b94504353733a2c@PROVIDERIP^M<br>
        CSeq: 103 INVITE^M<br>
        Server: Enswitch^M<br>
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
        NOTIFY, INFO, PUBLISH^M<br>
        Supported: replaces^M<br>
        Contact: <a class="moz-txt-link-rfc2396E" href="sip:PHONENUMBER@PROVIDERMEDIAIP:5060"><sip:PHONENUMBER@PROVIDERMEDIAIP:5060></a>^M<br>
        Content-Type: application/sdp^M<br>
        Content-Length: 286^M<br>
        ^M<br>
        v=0^M<br>
        o=root 2110894460 2110894461 IN IP4 PROVIDERMEDIAIP^M<br>
        s=Asterisk PBX 11.3.0^M<br>
        c=IN IP4 PROVIDERMEDIAIP^M<br>
        t=0 0^M<br>
        m=audio 15594 RTP/AVP 0 8 3 101^M<br>
        a=rtpmap:0 PCMU/8000^M<br>
        a=rtpmap:8 PCMA/8000^M<br>
        a=rtpmap:3 GSM/8000^M<br>
        a=rtpmap:101 telephone-event/8000^M<br>
        a=fmtp:101 0-16^M<br>
        a=ptime:20^M<br>
        a=sendrecv^M<br>
        <br>
        13:44:04.164519 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP,
        length: 525<br>
        <a class="moz-txt-link-abbreviated" href="mailto:E..)!A..@..v....g.v.......T.ACK">E..)!A..@..v....g.v.......T.ACK</a> <a class="moz-txt-link-freetext" href="sip:PHONENUMBER@PROVIDERIP:5060">sip:PHONENUMBER@PROVIDERIP:5060</a>
        SIP/2.0^M<br>
        Via: SIP/2.0/UDP
        172.21.0.226;branch=z9hG4bKfe94.472e9fc0479de79b4f176cc9585d8880.0^M<br>
        Via: SIP/2.0/UDP
        127.0.0.1:5080;branch=z9hG4bK752b5264;rport=5080^M<br>
        Route: <a class="moz-txt-link-rfc2396E" href="sip:PROVIDERIP;lr=on"><sip:PROVIDERIP;lr=on></a>^M<br>
        Max-Forwards: 69^M<br>
        From: "asterisk"
        <a class="moz-txt-link-rfc2396E" href="sip:PROVIDERUSER@PROVIDERIP:5080"><sip:PROVIDERUSER@PROVIDERIP:5080></a>;tag=as65919d92^M<br>
        To: <a class="moz-txt-link-rfc2396E" href="sip:PHONENUMBER@PROVIDERIP"><sip:PHONENUMBER@PROVIDERIP></a>;tag=as260fefaa^M<br>
        Contact: <a class="moz-txt-link-rfc2396E" href="sip:PROVIDERUSER@127.0.0.1:5080"><sip:PROVIDERUSER@127.0.0.1:5080></a>^M<br>
        Call-ID: 271ac7a174d613cd0b94504353733a2c@PROVIDERIP^M<br>
        CSeq: 103 ACK^M<br>
        User-Agent: Elastix 3.0^M<br>
        Content-Length: 0^M<br>
        <br>
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      <br>
      <pre wrap="">_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
<a class="moz-txt-link-abbreviated" href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a>
<a class="moz-txt-link-freetext" href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a>
</pre>
    </blockquote>
    <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>
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - <a class="moz-txt-link-freetext" href="http://www.kamailioworld.com">http://www.kamailioworld.com</a></pre>
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