[Freeswitch-users] Calls ending after around 5min

[Michael Collins]

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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>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
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>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
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>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
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