Ugh, sorry everyone. I did just address those details/attachments to Oz, but then gmail changed it back to being sent to the full list :-(<div><br clear="all">Cheers,<br>Fraser<br><br><br>
<br><br><div class="gmail_quote">On 15 February 2012 11:16, Fraser Redmond <span dir="ltr">&lt;<a href="mailto:fraserredmond@gmail.com">fraserredmond@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">

Thanks Oz, I&#39;ll take you up on that offer and send details and files in a separate email.<div><br></div><div>If anyone else can take a look let me know and I&#39;ll email it to you too.<br><br>
Cheers,<br>Fraser<br><br><br>
<br><br><div class="gmail_quote"><div class="im">On 15 February 2012 03:39, Oz Mortimer <span dir="ltr">&lt;<a href="mailto:oz@o-tec.tv" target="_blank">oz@o-tec.tv</a>&gt;</span> wrote:<br></div><div><div class="h5"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">


<div lang="EN-GB" 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">Hi Fraser,<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">Your console output in pastbin isn’t complete so of limited use. I’d really need to see you dialplan xml along with possibly your js – feel free to pm me. You dialplan seems overly complicated too, you should be able to all of that within a single js application.<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"><u></u> <u></u></span></p>


<div style="border:none;border-top:solid #b5c4df 1.0pt;padding:3.0pt 0cm 0cm 0cm"><p class="MsoNormal"><b><span lang="EN-US" style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span lang="EN-US" 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>Fraser Redmond<br>


<b>Sent:</b> 14 February 2012 21:26<br><b>To:</b> FreeSWITCH Users Help</span></p><div><br><b>Subject:</b> Re: [Freeswitch-users] Calls ending after around 5min<u></u><u></u></div><p></p></div><p class="MsoNormal">
<u></u> <u></u></p><p class="MsoNormal">I thought that too, and had done a text-search my code for sched_hangup, and there&#39;s no mention of it anywhere.<u></u><u></u></p><div><div><div><p class="MsoNormal"><u></u> <u></u></p>


</div><div><p class="MsoNormal">Is there anything else that would set it based on other variables being set? The closest things I have is I&#39;m setting api_hangup_hook to run some javascript when the call is hung up, and hangup_after_bridge=false<u></u><u></u></p>


</div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">My dialplan is fairly complicated (several js files get run to do database logging, business logic, etc.) So the console output is probably more helpful:<u></u><u></u></p>


</div><div><p class="MsoNormal"><a href="http://pastebin.freeswitch.org/18385" target="_blank">http://pastebin.freeswitch.org/18385</a> <u></u><u></u></p></div><div><p class="MsoNormal" style="margin-bottom:12.0pt"><br clear="all">


Cheers,<br>Fraser<br><br><br><br><u></u><u></u></p><div><p class="MsoNormal">On 14 February 2012 15:41, Michael Collins &lt;<a href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>&gt; wrote:<u></u><u></u></p>


<p class="MsoNormal">Like omortimer says it looks suspicious. I&#39;d pastebin your dialplan and the console log of a single call coming in (or going out) and let&#39;s see if you&#39;ve got a sched_hangup in there somewhere.<span style="color:#888888"><br>


<br><span>-MC</span></span><u></u><u></u></p><div><div><p class="MsoNormal" style="margin-bottom:12.0pt"><u></u> <u></u></p><div><p class="MsoNormal">On Tue, Feb 14, 2012 at 12:26 PM, Fraser Redmond &lt;<a href="mailto:fraserredmond@gmail.com" target="_blank">fraserredmond@gmail.com</a>&gt; wrote:<u></u><u></u></p>


<p class="MsoNormal">I&#39;ve done a test call, and got a pcap of it, along with a sip debug. (I can email you files if you are willing to take a look.)<u></u><u></u></p><div><p class="MsoNormal"><u></u> <u></u></p></div>


<div><p class="MsoNormal">There&#39;s no sip traffic until it gets to the log lines I pasted before, which is followed by the server sending a BYE to the gateway and client.<u></u><u></u></p></div><div><p class="MsoNormal">


<u></u> <u></u></p></div><div><p class="MsoNormal">I did a google search on &quot;nat timeout&quot;, but the only thing relevant seemed to be the rtp-timeout and session-timeout which I mentioned before.<u></u><u></u></p>


</div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">I really think there&#39;s a setting somewhere telling it to hangup. The logs have things like this as the call is being hung up (but before the BYE&#39;s are sent) which seem suspicious<u></u><u></u></p>


</div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">&gt; Added task 4 switch_ivr_schedule_hangup (uuid) to run at 1329249352<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p>


</div><div><p class="MsoNormal">&gt; ending bridge by request from read function<u></u><u></u></p></div><div><p class="MsoNormal" style="margin-bottom:12.0pt"><br clear="all">Cheers,<br>Fraser<br><br><br><br><u></u><u></u></p>


<div><p class="MsoNormal">On 14 February 2012 14:08, Michael Collins &lt;<a href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>&gt; wrote:<u></u><u></u></p><p class="MsoNormal" style="margin-bottom:12.0pt">


I would get pcaps of the calls that drop at 5 minutes, including media. If you need suggestions on how to collect pcaps then check out this page:<br><br><a href="http://wiki.freeswitch.org/wiki/Packet_Capture" target="_blank">http://wiki.freeswitch.org/wiki/Packet_Capture</a><br>


<br>Lots of tools at your disposal! Don&#39;t forget that Wireshark has a really nice interface for exploring SIP calls and RTP streams.<br><br>-MC<u></u><u></u></p><div><div><div><p class="MsoNormal">On Tue, Feb 14, 2012 at 10:16 AM, Fraser Redmond &lt;<a href="mailto:fraserredmond@gmail.com" target="_blank">fraserredmond@gmail.com</a>&gt; wrote:<u></u><u></u></p>


</div></div><blockquote style="border:none;border-left:solid #cccccc 1.0pt;padding:0cm 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm"><div><div><p class="MsoNormal">I&#39;ve got calls that are getting disconnected after 5 minutes (.<u></u><u></u></p>


<div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">There&#39;s some calls that don&#39;t, and we don&#39;t have many calls that last that long (which is why we hadn&#39;t noticed until now.)<u></u><u></u></p>


<div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">I&#39;ve had a look through the documentation, and the only thing I saw that looked likely was the rtp-timeouts which I changed to this without any change:<br>


    &lt;param name=&quot;rtp-timeout-sec&quot; value=&quot;1800&quot;/&gt;<u></u><u></u></p><div><p class="MsoNormal">    &lt;param name=&quot;rtp-hold-timeout-sec&quot; value=&quot;1800&quot;/&gt;<u></u><u></u></p></div>


<div><p class="MsoNormal">(changed in internal and external profiles)<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">Here&#39;s the log output for when the call ends:<u></u><u></u></p>


</div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><div><p class="MsoNormal">9d39d866-5732-11e1-b1a8-05fc5f64fa97 2012-02-14 17:42:58.340991 [DEBUG] switch_core_session.c:1009 Send signal sofia/internal/<a href="http://sip:12605@99.237.220.56:3907" target="_blank">sip:12605@99.237.220.56:3907</a> [BREAK]<u></u><u></u></p>


</div><div><p class="MsoNormal">2012-02-14 17:42:58.340991 [DEBUG] switch_scheduler.c:214 Added task 4 switch_ivr_schedule_hangup (9d39d866-5732-11e1-b1a8-05fc5f64fa97) to run at 1329241378<u></u><u></u></p></div><div><p class="MsoNormal">


9d39d866-5732-11e1-b1a8-05fc5f64fa97 2012-02-14 17:42:58.361011 [DEBUG] switch_core_session.c:726 Send signal sofia/internal/<a href="http://sip:12605@99.237.220.56:3907" target="_blank">sip:12605@99.237.220.56:3907</a> [BREAK]<u></u><u></u></p>


</div><div><p class="MsoNormal">9d39d866-5732-11e1-b1a8-05fc5f64fa97 2012-02-14 17:42:58.521032 [DEBUG] switch_ivr.c:589 sofia/internal/<a href="http://sip:12605@99.237.220.56:3907" target="_blank">sip:12605@99.237.220.56:3907</a> Command Execute respond(603)<u></u><u></u></p>


</div><div><p class="MsoNormal">9d39d866-5732-11e1-b1a8-05fc5f64fa97 EXECUTE sofia/internal/<a href="http://sip:12605@99.237.220.56:3907" target="_blank">sip:12605@99.237.220.56:3907</a> respond(603)<u></u><u></u></p></div>


<div><p class="MsoNormal">9d39d866-5732-11e1-b1a8-05fc5f64fa97 2012-02-14 17:42:58.521032 [DEBUG] mod_sofia.c:2363 Responding with 603 [Decline]<u></u><u></u></p></div><div><p class="MsoNormal"><br>Any other settings I should be looking at?<u></u><u></u></p>


</div></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal" style="margin-bottom:12.0pt">Cheers,<br>Fraser<br><br><u></u><u></u></p></div></div></div><p class="MsoNormal"><u></u> <u></u></p></div>


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