asterisk users - Jan 2010 - Sip Trunk takes incomming calls for 2 minutes and then stops

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.

do you use the

qualify=yes

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.
>
>