(at least this is what I had to do in Trixbox CE)

Login to freePBX administrative interface
Click on Setup
in top right of page
Click on Trunks
in left side navigation
Click Add SIP Trunk
in middle of page
Scroll to Outgoing
Settings and enter didforsale_did
into Trunk Name field

Copy and paste the following into the PEER Details field:

Peer Details:
type=peer
host=209.216.2.211
nat=no
canrinvite=no
disallow=all
allow=ulaw&alaw
dtmfmode=rfc2833
insecure=very
qualify=yes

Click on Submit Changes to add your new SIP trunk to your Asterisk server
Click on the red bar at the top of the screen to apply the changes you just made

in extensions.conf add thease lines

[custom-a2billing]
exten => _X.,1,Answer
exten => _X.,n,Wait(1)
exten => _X.,n,DeadAGI(a2billing.php|1)
exten => _X.,n,Hangup

in FreePBX UI create a Custom Destination;
custom-a2billing,${EXTEN},1
Description: A2Billing
Notes: For a call through service

Click on Inbound Routes
 
Click on Add Incoming
Route.
You will first want to fill the
DID Number field with your
DID
Make sure to leave the Caller ID Number and
Zaptel channel blank in order to match any
incoming call.
 This is useful if you wish to receive all calls
 
Scroll down to Set Destination
Choose A2Billing Custom Destinations
From Drop Down Menu
 
Click on Submit Changes to add your new
inbound route to your Asterisk server
 
Click on the red bar at the top of the screen to apply the changes you
just made