If you use Asterisk, then the configuration required on your server is quite straightforward.
In the relevant part of your Asterisk "extensions.conf" insert the following lines:
exten => [your_phone_number},1,Dial(SIP/201)
...replacing [your_phone_number] with the phone number you purchased on sign up.
This will normally need to go in your [default] context unless you have configured Asterisk to route inbound sip calls from "sip.gradwell.net" to another context.
You should replace the Dial(SIP/201) part with an Asterisk function to route the call to your phone or a number of phones. This example assumes your phone is logged into your Asterisk server as extension "201".
If you have purchased a block of numbers, repeat the above for each number.