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.)<div><br></div><div>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.</div>
<div><br></div><div>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.</div><div><br></div><div>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</div>
<div><br></div><div>> Added task 4 switch_ivr_schedule_hangup (uuid) to run at 1329249352</div><div><br></div><div>> ending bridge by request from read function</div><div><br clear="all">Cheers,<br>Fraser<br><br><br>
<br><br><div class="gmail_quote">On 14 February 2012 14:08, Michael Collins <span dir="ltr"><<a href="mailto:msc@freeswitch.org">msc@freeswitch.org</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
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<br><br><div class="gmail_quote"><div><div class="h5">On Tue, Feb 14, 2012 at 10:16 AM, Fraser Redmond <span dir="ltr"><<a href="mailto:fraserredmond@gmail.com" target="_blank">fraserredmond@gmail.com</a>></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">I've got calls that are getting disconnected after 5 minutes (.<div><br></div><div>
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.)<div>
<br></div><div>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"/><br>
<div> <param name="rtp-hold-timeout-sec" value="1800"/></div><div>(changed in internal and external profiles)</div><div><br></div><div>Here's the log output for when the call ends:</div><div>
<br></div><div><div>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]</div>
<div>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</div><div>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]</div>
<div>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)</div>
<div>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)</div><div>9d39d866-5732-11e1-b1a8-05fc5f64fa97 2012-02-14 17:42:58.521032 [DEBUG] mod_sofia.c:2363 Responding with 603 [Decline]</div>
<div><br>Any other settings I should be looking at?</div></div><div><br></div><div>Cheers,<br>Fraser<br><br><br>
</div></div></div>
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