[asterisk-users] Call terminates after 20 minutes

[John Todd]

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