In order to configure Asterisk to Gizmo and use this service to dial in and out of our local network, we had to complete four steps.

The following steps was completed to allow us to call to anywhere around the United States for free.

1. Configure Google Voice phone number to go directly to voicemail.
2. Confrigure Gizmo5 phone number to forward all calls and send them via SIP to our Google Voice phone number.
3. Configure FreePBX to route all calls with a GV prefix to our Gizmo SIP URI.
4. Configure Asterisk to jump through the autodialer to place an outbound call to any U.S. phone number through Google Voice telephone interface.

To begin, we had to set up a Google Voice account. We then logged into our Google Voice account and clicked on Settings > General. In the Voicemail Greeting section, we recorded our greeting. In the Notification section, we inputted the email and SMS addresses for delivery to our voicemail messages. In the Voicemail Transcripts, we checked the option to transcribe voicemails. Finally we clicked on the Do Not Disturb check box to forward all inbound calls to voicemail. By doing this will allow us to call our Google Voice phone number and press a few keys to make an outbound call instead of listening to our voicemails.

Next, we had to set up a Gizmo account. This is used to forward all incoming calls to the SIP URI of our Google Voice phone number. We can get Gizmo5 with a SIP call, but we can no longer place a direct SIP call to Google Voice. Gizmo can though. We configured Asterisk to place a SIP call to our Gizmo phone number. Gizmo passes the call along with a SIP call to Google Voice.

Asterisk will count to ten while the call is transferred to Google Voice. Asterisk acts like an autodialer by sending *, entering our Google Voice password, pressing 1, and finally dialing a 10-digit number plus # on the phone to place a free call to anywhere in the U.S.

The SIP call from our Asterisk server to our Gizmo number is free. The SIP connection from Gizmo transferring the call to our Google Voice number is free. Any phone call to any number in the U.S. through Google Voice is free.

Sip.conf

[proxy01.sipphone.com]
context = default
type = peer
disallow = all
allow = ulaw,alaw,g726,speex,ilbc,gsm,lpc10,g729
dtmfmode = rfc2833
host = proxy01.sipphone.com
fromdomain = proxy01.sipphone.com
insecure = very
qualify = yes
fromuser = 17475038061
authuser = 17475038061
username = 17475038061
secret = EnterPasswordHere
canreinvite = no

• Comments on the code

o The [proxy01.sipphone.com] section defines a peer from which asterisk will accept SIP INVITES.

o fromuser = 174750338061  when the person calls out it will act like an address resolution protocol (ARP). This means it tells the system that this particular person is from this computer; and starts the authentication process. When the person calls out they will be required to go through 3 steps.

1. Gives you a username.
2. Authenticates what username it is.
3. Tells from what user.

o dtmfmode = rfc2833  signaling method to carry trunk signals.

o allow = ulaw,alaw,g726,speex,ilbc,gsm,lpc10,g729  digital representation of an analog signal where the magnitude of the signal is sampled regularly at uniform intervals.

Extensions.conf

[outbound]

exten => _1X.,1,SetCallerID("John Doe" <17475038061>)
exten => _1X.,2,Dial(SIP/${EXTEN:1}@proxy01.sipphone.com,20,r)
exten => _1X.,3,Hangup