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    <div class="moz-cite-prefix"><tt>Hello Jorge,<br>
        <br>
        Just as RFC3261 says, 1XX responses create "early dialogs",
        while 2XX final responses create "dialogs". OpenSIPS follows the
        same rules, only that it considers that the "true" lifetime of a
        dialog begins when the callee picks up his/her phone. That is
        why those fields are still unused - they are initialized to
        zero, since the 200 OK is still to come.<br>
        <br>
        If that dialog were to remain in the early state, the
        transaction would expire eventually, triggering a delete of that
        early dialog.<br>
        <br>
        Best regards,<br>
      </tt>
      <pre class="moz-signature" cols="72">Liviu Chircu
OpenSIPS Developer
<a class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a></pre>
      On 05/21/2014 07:23 PM, Jorge Ortea wrote:<br>
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    <blockquote cite="mid:DUB128-W82BAE5A3126EA98349A8CFDC3C0@phx.gbl"
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      <div dir="ltr">Hi all,<br>
        <br>
        when I execute 'opensipsctl fifo dlg_list' shows this:<br>
        <br>
        dialog::&nbsp; hash=206:862937179<br>
        &nbsp;&nbsp;&nbsp; state:: 2<br>
        &nbsp;&nbsp;&nbsp; user_flags:: 0<br>
        &nbsp;&nbsp;&nbsp; timestart:: 0<br>
        &nbsp;&nbsp;&nbsp; timeout:: 0<br>
        &nbsp;&nbsp;&nbsp; callid:: <a class="moz-txt-link-abbreviated" href="mailto:786344563@A.B.C.D">786344563@A.B.C.D</a><br>
        &nbsp;&nbsp;&nbsp; from_uri:: sip:xxxxxxxxx@MyDomain<br>
        &nbsp;&nbsp;&nbsp; to_uri:: sip:yyyyyyyyy@MyDomain<br>
        &nbsp;&nbsp;&nbsp; caller_tag:: 356538242<br>
        &nbsp;&nbsp;&nbsp; caller_contact:: <a class="moz-txt-link-abbreviated" href="mailto:sip:xxxxxxxx@A.B.C.D:5062">sip:xxxxxxxx@A.B.C.D:5062</a><br>
        &nbsp;&nbsp;&nbsp; callee_cseq:: 2<br>
        &nbsp;&nbsp;&nbsp; caller_route_set:: <br>
        &nbsp;&nbsp;&nbsp; caller_bind_addr:: udp:MyDomain<br>
        &nbsp;&nbsp;&nbsp; callee_tag:: as28f3b5f8<br>
        &nbsp;&nbsp;&nbsp; callee_contact:: <br>
        &nbsp;&nbsp;&nbsp; caller_cseq:: 2<br>
        &nbsp;&nbsp;&nbsp; callee_route_set:: <br>
        &nbsp;&nbsp;&nbsp; callee_bind_addr:: udp:MyDomain<br>
        <br>
        <br>
        This segment refered to a call from UAC to a mediagateway
        through Proxy SIP (Opensips 1.6.4-2-tls)<br>
        <br>
        Is it normal that both timestart and timeout are 0?<br>
        <br>
        Thanks.<br>
        Regards.<br>
      </div>
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</pre>
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