Thanks Oz, I'll take you up on that offer and send details and files in a separate email. If anyone else can take a look let me know and I'll email it to you too. Cheers, Fraser On 15 February 2012 03:39, Oz Mortimer <oz@o-tec.tv> wrote:
Hi Fraser,****
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.****
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*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Fraser Redmond *Sent:* 14 February 2012 21:26 *To:* FreeSWITCH Users Help
*Subject:* Re: [Freeswitch-users] Calls ending after around 5min****
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I thought that too, and had done a text-search my code for sched_hangup, and there's no mention of it anywhere.****
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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****
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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:****
http://pastebin.freeswitch.org/18385 ****
Cheers, Fraser
****
On 14 February 2012 15:41, Michael Collins <msc@freeswitch.org> wrote:****
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.
-MC****
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On Tue, Feb 14, 2012 at 12:26 PM, Fraser Redmond <fraserredmond@gmail.com> wrote:****
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.)****
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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.** **
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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.****
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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****
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Added task 4 switch_ivr_schedule_hangup (uuid) to run at 1329249352****
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ending bridge by request from read function****
Cheers, Fraser
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On 14 February 2012 14:08, Michael Collins <msc@freeswitch.org> wrote:****
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:
http://wiki.freeswitch.org/wiki/Packet_Capture
Lots of tools at your disposal! Don't forget that Wireshark has a really nice interface for exploring SIP calls and RTP streams.
-MC****
On Tue, Feb 14, 2012 at 10:16 AM, Fraser Redmond <fraserredmond@gmail.com> wrote:****
I've got calls that are getting disconnected after 5 minutes (.****
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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.)****
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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: <param name="rtp-timeout-sec" value="1800"/>****
<param name="rtp-hold-timeout-sec" value="1800"/>****
(changed in internal and external profiles)****
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Here's the log output for when the call ends:****
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9d39d866-5732-11e1-b1a8-05fc5f64fa97 2012-02-14 17:42:58.340991 [DEBUG] switch_core_session.c:1009 Send signal sofia/internal/ sip:12605@99.237.220.56:3907 [BREAK]****
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****
9d39d866-5732-11e1-b1a8-05fc5f64fa97 2012-02-14 17:42:58.361011 [DEBUG] switch_core_session.c:726 Send signal sofia/internal/ sip:12605@99.237.220.56:3907 [BREAK]****
9d39d866-5732-11e1-b1a8-05fc5f64fa97 2012-02-14 17:42:58.521032 [DEBUG] switch_ivr.c:589 sofia/internal/sip:12605@99.237.220.56:3907 Command Execute respond(603)****
9d39d866-5732-11e1-b1a8-05fc5f64fa97 EXECUTE sofia/internal/ sip:12605@99.237.220.56:3907 respond(603)****
9d39d866-5732-11e1-b1a8-05fc5f64fa97 2012-02-14 17:42:58.521032 [DEBUG] mod_sofia.c:2363 Responding with 603 [Decline]****
Any other settings I should be looking at?****
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Cheers, Fraser
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org****
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org****
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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