Getting the sip_call_id variable (Call-ID header) for each leg (eg from xml cdrs) will give you a field that makes it very easy to pull the call out of a pcap.<div><br></div><div>-Steve</div><div><br></div><div><br><br><div class="gmail_quote">

On 18 December 2012 00:00, Anthony Minessale <span dir="ltr">&lt;<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

<div dir="ltr">usually the 406 is done in an established call to refuse a codec change during re-invite.<div>Its possible the other end thinks we want to change the codec when we do the session-timer re-invite but I&#39;m sure we don&#39;t but the sip trace can help shed some light.  You can run a pcap too at the same time so when we find the bad call in the logs we can filter it out of the pcap too.  To avoid it getting too big you can just restart it every so often or use sippcapdump and delete calls that are not affected.</div>


<div><br></div><div><br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote"><div><div class="h5">On Mon, Dec 17, 2012 at 5:51 PM, Sean Devoy <span dir="ltr">&lt;<a href="mailto:sdevoy@bizfocused.com" target="_blank">sdevoy@bizfocused.com</a>&gt;</span> wrote:<br>


</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div class="h5"><div lang="EN-US" link="blue" vlink="purple"><div><p class="MsoNormal"><span style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">Waiting for another failure with siptrace REALLY on this time.<u></u><u></u></span></p>


<p class="MsoNormal"><span style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">If the user has clicked DND on these cisco phones, could that cause these messages?<u></u><u></u></span></p>


<p class="MsoNormal"><span style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">Sean<u></u><u></u></span></p>


<p class="MsoNormal"><span style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a> [mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a>] <b>On Behalf Of </b>Anthony Minessale<br>


<b>Sent:</b> Monday, December 17, 2012 12:32 PM</span></p><div><br><b>To:</b> FreeSWITCH Users Help<br><b>Subject:</b> Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]<u></u><u></u></div><p></p>
<p class="MsoNormal"><u></u> <u></u></p><div><p class="MsoNormal">The siptrace is not in here?  Did you enable it on the console and then capture to the log file perhaps?<u></u><u></u></p></div><div><div><div><p class="MsoNormal">


you should do &quot;sofia tracelevel debug&quot; too to route traces to the log file.<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">Its hard to say for sure with no sip trace but it seems like the far end is rejecting the session timer re-invite causing the call to end.<u></u><u></u></p>


</div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p><div><p class="MsoNormal">On Mon, Dec 17, 2012 at 11:05 AM, Sean Devoy &lt;<a href="mailto:sdevoy@bizfocused.com" target="_blank">sdevoy@bizfocused.com</a>&gt; wrote:<u></u><u></u></p>


<p class="MsoNormal"><a href="http://pastebin.freeswitch.org/20342" target="_blank">http://pastebin.freeswitch.org/20342</a><u></u><u></u></p></div><p class="MsoNormal"><br><br clear="all"><u></u><u></u></p><div><p class="MsoNormal">


<u></u> <u></u></p></div><p class="MsoNormal">-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/" target="_blank">http://www.cluecon.com/</a><br>


Twitter: <a href="http://twitter.com/FreeSWITCH_wire" target="_blank">http://twitter.com/FreeSWITCH_wire</a><br><br>AIM: anthm<br><a href="mailto:MSN%3Aanthony_minessale@hotmail.com" target="_blank">MSN:anthony_minessale@hotmail.com</a><br>


GTALK/JABBER/<a href="mailto:PAYPAL%3Aanthony.minessale@gmail.com" target="_blank">PAYPAL:anthony.minessale@gmail.com</a><br>IRC: <a href="http://irc.freenode.net" target="_blank">irc.freenode.net</a> #freeswitch<br><br>

FreeSWITCH Developer Conference<br>
<a href="mailto:sip%3A888@conference.freeswitch.org" target="_blank">sip:888@conference.freeswitch.org</a><br><a href="mailto:googletalk%3Aconf%2B888@conference.freeswitch.org" target="_blank">googletalk:conf+888@conference.freeswitch.org</a><br>


pstn:<a href="tel:%2B19193869900" value="+19193869900" target="_blank">+19193869900</a><u></u><u></u></p></div></div></div></div></div><br></div></div><div class="im">_________________________________________________________________________<br>


Professional FreeSWITCH Consulting Services:<br>
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<br>
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<br>
Official FreeSWITCH Sites<br>
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<br></div></blockquote></div><br><br clear="all"><div class="im"><div><br></div>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/" target="_blank">http://www.cluecon.com/</a><br>


Twitter: <a href="http://twitter.com/FreeSWITCH_wire" target="_blank">http://twitter.com/FreeSWITCH_wire</a><br><br>AIM: anthm<br><a href="mailto:MSN%3Aanthony_minessale@hotmail.com" target="_blank">MSN:anthony_minessale@hotmail.com</a><br>

GTALK/JABBER/<a href="mailto:PAYPAL%3Aanthony.minessale@gmail.com" target="_blank">PAYPAL:anthony.minessale@gmail.com</a><br>
IRC: <a href="http://irc.freenode.net" target="_blank">irc.freenode.net</a> #freeswitch<br><br>FreeSWITCH Developer Conference<br><a href="mailto:sip%3A888@conference.freeswitch.org" target="_blank">sip:888@conference.freeswitch.org</a><br>

<a href="mailto:googletalk%3Aconf%2B888@conference.freeswitch.org" target="_blank">googletalk:conf+888@conference.freeswitch.org</a><br>
pstn:<a href="tel:%2B19193869900" value="+19193869900" target="_blank">+19193869900</a><br>
</div></div>
<br>_________________________________________________________________________<br>
Professional FreeSWITCH Consulting Services:<br>
<a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br>
<a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a><br>
<br>
FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>
<a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a><br>
<br>
Official FreeSWITCH Sites<br>
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<br></blockquote></div><br></div>