Configure sip trunk GNR

[baritello]

Should I set up an Asterisk PBX with a service purchased by the supplier Colt Italy, VoIP access with range 100 numbers.
Pending a technical advice that we will soon, I wanted to begin testing the line, and I've already downloaded and installed a virtual server carrying the latest version of AsteriskNow. Someone could give me some guidance to configure the Trunk sip to DDI and the rules of entry and exit to handle 100 numbers with some internal sample?
Thank you all

[comfine]

for handling "inbound route" for multiple numbers we always write a couple of lines to extensions_custom.conf

for example:

[from-trunk]
exten => s,1,Ringing
exten => s,n,Answer(1000)
exten => s,n,Playback(/var/lib/asterisk/sounds/custom/welcome)
exten => s,n,Set(thedid=${CALLERID(dnid)})
exten => s,n,Set(DIAL_OPTIONS=m)
exten => s,n,Goto(from-internal,${thedid:-2},1)

this code will answer the line, playback some welcomemessage, and then the last two numbers of the called number will be cut out and used as dial for the internal extension, while the caller hears some music meanwhile....

if you dial 01234-110 it will redirect the call to extension 10
if you dial 01234-120 it will redirect the call to extension 20
if you dial 01234-144 it will redirect the call to extension 44

i don't know if there's a more comfortable way in freepbx - i never found one....

the trunk-settings depends on the provider. in most cases you have to fill-out the registration-field in freepbx!