[asterisk-users] Sip Trunk takes incomming calls for 2 minutes and then stops

[Yves Arikoglu]

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.
>
>