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--></style></head><body lang="EN-US" link="blue" vlink="purple"><div class="WordSection1"><p class="MsoNormal"><span style="color:#1F497D">Please ignore – it appears it’s due to the “Session-Expires: -1” from Asterisk – that’s causing FS to drop the 200 OK.</span></p>
<p class="MsoNormal"><span style="color:#1F497D"> </span></p><p class="MsoNormal"><span style="color:#1F497D">Thanks,</span></p><p class="MsoNormal"><span style="color:#1F497D"> </span></p><div><p class="MsoNormal"><span style="color:#1F497D">Blake</span></p>
</div><p class="MsoNormal"><span style="color:#1F497D"> </span></p><div><div style="border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in"><p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> Blake Sevier [mailto:<a href="mailto:bsevier@bandwidth.com">bsevier@bandwidth.com</a>] <br>
<b>Sent:</b> Wednesday, May 25, 2011 10:58 AM<br><b>To:</b> '<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>'<br><b>Subject:</b> FS not recognizing 200 OK after far-end call is answered</span></p>
</div></div><p class="MsoNormal"> </p><p class="MsoNormal">Hi all, </p><p class="MsoNormal"> </p><p class="MsoNormal">I’m new, so please bear with me. My setup is fairly basic, and the routing all seems to work just fine:</p>
<p class="MsoNormal"> </p><p class="MsoNormal">FS receives call from an OpenSIPS proxy, it answers the call, records it and plays back a short prompt before invoking the “bridge” function to place an outbound call via the same OpenSIPS proxy to an external SIP endpoint (in testing, just an Asterisk instance that answers the call). This all works just fine, except FS doesn’t seem to recognize the 200 OK it receives back for leg b after the Asterisk instances answers the call. It apparently does recognize/accept the 100 Trying, as FS doesn’t send additional INVITEs. I’m not seeing any issues with the SIP messaging received by FS – it’s apparently formatted correctly. </p>
<p class="MsoNormal">To note, I’ve also tested calls direct to the Asterisk instance (bypassing OpenSIPS proxy), and get similar results. My guess is that I’ve got something incorrectly configured within my dialplan?? However, not sure how to debug any further to figure out why the 200 Oks are not being recognized. Eventually, after 60s, FS ends the call with a NO_ANSWER.</p>
<p class="MsoNormal">Below is a link to a full call trace from FS – from start of call to when I see multiple 200 OKs from Asterisk.</p><p class="MsoNormal"><a href="http://pastebin.com/bmEtiW9X">http://pastebin.com/bmEtiW9X</a></p>
<p class="MsoNormal"> </p><p class="MsoNormal">Any help would be greatly appreciated. If any further details are needed, please let me know</p><p class="MsoNormal"> </p><p class="MsoNormal">Thank you!!!</p><p class="MsoNormal">
</p><p class="MsoNormal">Blake</p><p class="MsoNormal"> </p></div></body></html>