<html xmlns:v="urn:schemas-microsoft-com:vml" xmlns:o="urn:schemas-microsoft-com:office:office" xmlns:w="urn:schemas-microsoft-com:office:word" xmlns:m="http://schemas.microsoft.com/office/2004/12/omml" xmlns="http://www.w3.org/TR/REC-html40"><head><meta http-equiv=Content-Type content="text/html; charset=us-ascii"><meta name=Generator content="Microsoft Word 12 (filtered medium)"><style><!--
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</o:shapelayout></xml><![endif]--></head><body lang=EN-GB link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Hi Fraser,<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";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.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></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:"Tahoma","sans-serif"'>From:</span></b><span lang=EN-US style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] <b>On Behalf Of </b>Fraser Redmond<br><b>Sent:</b> 14 February 2012 21:26<br><b>To:</b> FreeSWITCH Users Help<br><b>Subject:</b> Re: [Freeswitch-users] Calls ending after around 5min<o:p></o:p></span></p></div><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>I thought that too, and had done a text-search my code for sched_hangup, and there's no mention of it anywhere.<o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></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'm setting api_hangup_hook to run some javascript when the call is hung up, and hangup_after_bridge=false<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></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:<o:p></o:p></p></div><div><p class=MsoNormal><a href="http://pastebin.freeswitch.org/18385">http://pastebin.freeswitch.org/18385</a> <o:p></o:p></p></div><div><p class=MsoNormal style='margin-bottom:12.0pt'><br clear=all>Cheers,<br>Fraser<br><br><br><br><o:p></o:p></p><div><p class=MsoNormal>On 14 February 2012 15:41, Michael Collins <<a href="mailto:msc@freeswitch.org">msc@freeswitch.org</a>> wrote:<o:p></o:p></p><p class=MsoNormal>Like omortimer says it looks suspicious. I'd pastebin your dialplan and the console log of a single call coming in (or going out) and let's see if you've got a sched_hangup in there somewhere.<span style='color:#888888'><br><br><span class=hoenzb>-MC</span></span><o:p></o:p></p><div><div><p class=MsoNormal style='margin-bottom:12.0pt'><o:p> </o:p></p><div><p class=MsoNormal>On Tue, Feb 14, 2012 at 12:26 PM, Fraser Redmond <<a href="mailto:fraserredmond@gmail.com" target="_blank">fraserredmond@gmail.com</a>> wrote:<o:p></o:p></p><p class=MsoNormal>I'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.)<o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>There'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.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>I did a google search on "nat timeout", but the only thing relevant seemed to be the rtp-timeout and session-timeout which I mentioned before.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>I really think there'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's are sent) which seem suspicious<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>> Added task 4 switch_ivr_schedule_hangup (uuid) to run at 1329249352<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>> ending bridge by request from read function<o:p></o:p></p></div><div><p class=MsoNormal style='margin-bottom:12.0pt'><br clear=all>Cheers,<br>Fraser<br><br><br><br><o:p></o:p></p><div><p class=MsoNormal>On 14 February 2012 14:08, Michael Collins <<a href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>> wrote:<o:p></o:p></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't forget that Wireshark has a really nice interface for exploring SIP calls and RTP streams.<br><br>-MC<o:p></o:p></p><div><div><div><p class=MsoNormal>On Tue, Feb 14, 2012 at 10:16 AM, Fraser Redmond <<a href="mailto:fraserredmond@gmail.com" target="_blank">fraserredmond@gmail.com</a>> wrote:<o:p></o:p></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've got calls that are getting disconnected after 5 minutes (.<o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>There's some calls that don't, and we don't have many calls that last that long (which is why we hadn't noticed until now.)<o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>I'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> <param name="rtp-timeout-sec" value="1800"/><o:p></o:p></p><div><p class=MsoNormal> <param name="rtp-hold-timeout-sec" value="1800"/><o:p></o:p></p></div><div><p class=MsoNormal>(changed in internal and external profiles)<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Here's the log output for when the call ends:<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></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]<o:p></o:p></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<o:p></o:p></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]<o:p></o:p></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)<o:p></o:p></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)<o:p></o:p></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]<o:p></o:p></p></div><div><p class=MsoNormal><br>Any other settings I should be looking at?<o:p></o:p></p></div></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal style='margin-bottom:12.0pt'>Cheers,<br>Fraser<br><br><o:p></o:p></p></div></div></div><p class=MsoNormal><o:p> </o:p></p></div></div><p class=MsoNormal style='margin-bottom:12.0pt'>_________________________________________________________________________<br>Professional FreeSWITCH Consulting Services:<br><a href="mailto:consulting@freeswitch.org" target="_blank">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><a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br><a href="http://wiki.freeswitch.org" target="_blank">http://wiki.freeswitch.org</a><br><a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a><br><br>FreeSWITCH-users mailing list<br><a 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style='margin-bottom:12.0pt'><br>_________________________________________________________________________<br>Professional FreeSWITCH Consulting Services:<br><a href="mailto:consulting@freeswitch.org" target="_blank">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><a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br><a href="http://wiki.freeswitch.org" target="_blank">http://wiki.freeswitch.org</a><br><a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a><br><br>FreeSWITCH-users mailing list<br><a href="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank">FreeSWITCH-users@lists.freeswitch.org</a><br><a 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