≡ Asterisk Setup

I picked up a VTWhite account recently for five bucks - if you just want to toy around with SIP this is a great provider to go with. They offer DIDs for 75 cents a month and you only pay for the calls you make - so you don’t need to invest $25 bucks a month to go with a full fledged VoIP account.

VTWhite is a barebones trunk. It doesn’t have any features like voicemail, forwarding, or simrings - you have to code that stuff yourself. This means the configs are a little lighter in some areas, and a little longer in others.

An important note before we dive in: You do not register to VTWhite’s SIP peer. They send stuff to you based on IP, not a user. I’ve had a few people ask me how do you register, so hopefully that will save someone some time.

Open your sip.conf and copy/paste the following:

[vtwhite]
type=friend
insecure=very
host=sip.vtwhite.com
qualify=3600
nat=no

That’s it. Do a sip reload and then a sip show peers and you should see a connection between you and vtwhite.

Now, lets move away from Asterisk and make sure everything is setup right on VTWhite’s end. Log into the control panel and click on node manager. If you haven’t already, add your PBX’s details. The name can be anything you want - its internal to the CP to help you organize nodes. Source IP is the IP of your PBX. Destination hostname is your dns name. Destination port is what you’ve set Asterisk to use - it should be 5060. If you have any reason for it not to be, you probably shouldn’t be using VTWhite. This is for static IP, hosted PBXs. Use DHCP and dyndns at your own risk.

With the node setup, we can add DIDs to it. Click on Add Number and choose an area code or state. Scroll through the list and choose your number. Notice the dropdown with your Node in it. There’s only 1 now, but as you add nodes you can point specific numbers to each. Click Get Number.

If you get an error, you might not be verified, or you need to add money to your account. Otherwise you should see your new number.

Take that number, and create a peer in sip.conf for your phone.

[15553331111]
type=friend
host=dynamic
callerid="VTWhite" <15553331111>
username=15553331111
secret=password
nat=yes
canreinvite=yes
disallow=all
allow=ulaw
context=vtwhite

Next, we need to setup routes for the new DID. We’re going to structure this so that you can easily add another peer to sip.conf without having to change the routes.

[vtwhite]
include=vtwhite-outgoing
include=vtwhite-incoming

[vtwhite-outgoing]
; Make sure we pass an 11 digit number to vtwhite's SIP proxy
exten => _NXXXXXX,1,Goto(1${CALLERID(num):1:3}${EXTEN},1)
exten => _NXXNXXXXXX,1,Goto(1${EXTEN},1)

exten => _1NXXNXXXXXX,1,Verbose(1| ${CALLERID(num)} is calling ${EXTEN})
exten => _1NXXNXXXXXX,2,Dial(SIP/${EXTEN}@vtwhite,40,r)
exten => _1NXXNXXXXXX,3,Hangup

[vtwhite-incoming]
; Answer the call and strip the number dialed from the sip header
exten => s,1,Answer
exten => s,2,Set(VTWDID=${SIP_HEADER(TO):5:11})
exten => s,3,Goto(${VTWDID},1)

exten => _1NXXNXXXXXX,1,Verbose(1|Incoming VTWHITE call for ${EXTEN})
exten => _1NXXNXXXXXX,2,Dial(SIP/${EXTEN},25,r)
exten => _1NXXNXXXXXX,3,Hangup