[Freeswitch-users] Calls ending after around 5min

Wed Feb 15 19:16:55 MSK 2012

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 at 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.****
>
> ** **
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at 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****
>
> ** **
>
> I thought that too, and had done a text-search my code for sched_hangup,
> and there's no mention of it anywhere.****
>
> ** **
>
> 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****
>
> ** **
>
> 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 at 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****
>
> ** **
>
> On Tue, Feb 14, 2012 at 12:26 PM, Fraser Redmond <fraserredmond at 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.)****
>
> ** **
>
> 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.**
> **
>
> ** **
>
> 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.****
>
> ** **
>
> 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****
>
> ** **
>
> > Added task 4 switch_ivr_schedule_hangup (uuid) to run at 1329249352****
>
> ** **
>
> > ending bridge by request from read function****
>
>
> Cheers,
> Fraser
>
>
>
> ****
>
> On 14 February 2012 14:08, Michael Collins <msc at 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 at gmail.com>
> wrote:****
>
> I've got calls that are getting disconnected after 5 minutes (.****
>
> ** **
>
> 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.)****
>
> ** **
>
> 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)****
>
> ** **
>
> Here's the log output for when the call ends:****
>
> ** **
>
> 9d39d866-5732-11e1-b1a8-05fc5f64fa97 2012-02-14 17:42:58.340991 [DEBUG]
> switch_core_session.c:1009 Send signal sofia/internal/
> sip:12605 at 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 at 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 at 99.237.220.56:3907 Command
> Execute respond(603)****
>
> 9d39d866-5732-11e1-b1a8-05fc5f64fa97 EXECUTE sofia/internal/
> sip:12605 at 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?****
>
> ** **
>
> Cheers,
> Fraser
>
> ****
>
> ** **
>
> _________________________________________________________________________
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>
> 
> 
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>
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>
> 
> 
>
> Official FreeSWITCH Sites
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> ** **
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
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>
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>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
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>
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>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
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