[asterisk-users] Sip Trunk takes incomming calls for 2 minutes and then stops
Peter Childs
pchilds at bcs.org
Tue Jan 26 04:30:54 CST 2010
2010/1/26 Yves Arikoglu <yves030 at gmx.de>:
> do you use the
>
> qualify=yes
>
No, If I do it does not work at all.
I've found if I set defaultexpiry to 30 it works fine. and was infact
working for 30 seconds every two minutes before, It looks like
sipgate.co.uk are expiring there registry attempts very quickly.
Peter
> option for your endpoints?
>
> y.
>
>
> Peter Childs schrieb:
>> Using sipgate.co.uk, Asterisk, FreePBX and Asterisk in a Flash
>>
>> I've managed to get a basic system set up. and can now take and make
>> sip calls over the sip trunk I've got from sipgate.co.uk for testing
>> purposes....
>>
>> Anyway I can make calls fine (if only to the testing line and other
>> sipgate lines as I have not set up any credit), and I can take calls
>> but only if someone phones me within 2 minutes of doing a "sip reload"
>> otherwise I just get a dead line.
>>
>> I'm thinking this is something to do with registration or Nat, but
>> I've set my Nat up to forward everything, and it all works for
>> 2minutes.....
>>
>>
>>
>> Peter.
>>
>>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
More information about the asterisk-users
mailing list