[asterisk-users] Call terminates after 20 minutes

[John Todd]

Other things to check:

  - ensure you don't have media timeouts set (see "rtptimeout" in  
sip.conf)
  - do you have any absolute timers set in  your dialplan? (any $ 
{TIMEOUT(absolute)} values set?
  - does your upstream carrier have any sort of timer limit? (try  
using just a vanilla softphone to call directly to your trunking  
provider, no Asterisk involved.)

What does your console say in full debug mode?

JT



On Nov 4, 2008, at 8:04 AM, Jim Boykin wrote:

> Hi,
>
> It does not appears to be a session-timers issue. There are no SIP
> exchange except for BYE message initited after 20 minutes.
>
> I increased the session timer to 3600 seconds and also tried your
> suggestion without any luck. Any other inputs?
>
> Thanks
> Jim
>
>
>
> On Mon, Nov 3, 2008 at 10:47 PM, John Todd <jtodd at digium.com> wrote:
>>
>> Go to sip.conf.
>>
>> Find the SIP Session-Timers section.
>>
>> Ensure that you have this option set:
>>
>> session-timers=refuse
>>
>> This might help.  If not, try other variations of the session-timers
>> value.  The default session-timer is 10 minutes - exactly half of  
>> what
>> you claim is your duration maximums, so it seems suspiciously like
>> that might have something to do with it.  Maybe not.  In any case,
>> fire up wireshark/tethereal and watch the SIP packets for a  
>> particular
>> call to see what's happening - distrust everything other than what  
>> you
>> see on "the wire" and then work backwards.  An understanding of SIP
>> packet flows will be helpful here, or the "ladder view" of SIP
>> transactions that is built into  wireshark's graphical interface will
>> certainly help as well.
>>
>> JT
>>
>>
>> On Nov 2, 2008, at 11:07 AM, Jim Boykin wrote:
>>
>>> Any help. Thanks
>>>
>>>
>>> On Sun, Nov 2, 2008 at 12:50 PM, Jim Boykin <boykinjim at gmail.com>
>>> wrote:
>>>> Marcin, can you elaborate. No timer has been set and call is not
>>>> idle either.
>>>>
>>>> Thanks
>>>> Jim
>>>>
>>>> On Fri, Oct 31, 2008 at 5:17 PM, Marcin J. Kowalczyk
>>>> <marcin.kowalczyk at ccig.pl> wrote:
>>>>> Jim Boykin pisze:
>>>>>> We are running Asterisk SVN. We are facing a strange and  
>>>>>> repetable
>>>>>> problem. All outgoing call gets terminated in approx 20 minutes.
>>>>>> Asterisk initiates BYE message to the remote end and call
>>>>>> terminates.
>>>>>>
>>>>> Sesion-timer set but not supported by sip-peers?
>>>>
>>
>> ---
>> John Todd
>> jtodd at digium.com        +1-256-428-6083
>> Asterisk Open Source Community Director
>>
>>
>>
>>
>>
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---
John Todd
jtodd at digium.com        +1-256-428-6083
Asterisk Open Source Community Director