Cannot figure out how to add a DID SIP line

Can someone tell guide me how to configure a DID line in the Asterisk 2.0 Gui? I am able to register the Turnk with the provider, however, when I dial the number I get a busy signal and a message from my provider stating:

[i]A call to your DID 847603000 has failed at 8:17pm on 04/08/2009 MDT. We received ‘CONGESTION’ when attempting to route the call to your server or device.

This error usually means your server or device does not recognize the number being dialed. If using asterisk, make sure you have the correct inbound context specified on your inbound trunk and that you have correctly added an inbound route/extension logic for this DID.[/i]

I am lost with routing the line to the extension, some guidance would be appreciated.

Thank you,

I believe there is an Incoming call rules section in the GUI. You would need to put in 847603000 as an incoming caller route.

Your response was quick and easy. I was looking into the whole thing way to much trying to setup some rules in order to make it work.

Is there an easy solution for the outbound calls?

Also, is there a reference somewhere I can use to better understand the extensions and features?

Thank you

I am sure there is a manual some where for the GUI but I am not sure where. You may want to read “the book”: