Below is a screen shot of my trunks. Under the general settings under outbound caller id. This is where you enter the phone number that you want to show up on the caller id. Click on this picture to enlarge so you can see all the settings.
Here is a screenshot of my routes. There are a few things to keep in mind here. For your dial patterns enter the phone number that you will be calling with the spoofed caller id number. You may want to enter it in all of the following formats.
19999999999
9999999999
9999999
This way when you dial the number that you are trying to make a victim of the spoof it knows to use the VoicePulse route.
Manual Configuration of Voice Pulse
(Updated Aug 20, 2008)
If you have a packaged version of FreePBX (trixbox, PBX-in-a-Flash, etc) it is highly recommended that you use the FreePBX module from the section above.
Print this page for reference before you start
Login to your server via the web interface using a browser
Click on Trunks > Add SIP Trunk
Outgoing CallerID: 0000000000 (10-digits only) The name you set here will NOT be sent when you call regular PSTN lines.
Maximum Channels: Enter the number of channels you have purchased, 4 by default
If you are closer to San Jose, CA, use “sjc” instead of “jfk” (New York, NY) in the settings below.
Dial Rules:
011|.
1NXXNXXXXXX
1+NXXNXXXXXX
1732+NXXXXXX ;<– Replace 732 with your area code
Outbound Dial Prefix: +
Trunk Name: VP-SIPJFKA
Peer Details:
type=peer
host=jfk-primary.voicepulse.com
qualify=5000
allow=all
canreinvite=no
username=Your Login from the Credentials Page
secret=Your Password from the Credentials Page
dtmfmode=rfc2833
rfc2833compensate=yes
insecure=port,invite
trustrpid=yes
User Context: leave blank
User Details: leave blank
Register String: Login:Password@jfk-primary.voicepulse.com (use the Login and Password from the Credentials page)
Click “Submit Changes”
Repeat steps 4-14, except use VP-SIPJFKB and jfk-backup.voicepulse.com in place of VP-SIPJFKA and jfk-primary.voicepulse.com. You now have redundant trunks to VoicePulse!
Click on Outbound Routes > Add Route
Route Name: VP-OUT
Dial Patterns: Insert pre-defined patterns for Toll-free, Long Distance, and International
Click “Submit Changes”. You now have an outbound route to VoicePulse which will try both trunks defined earlier for toll-free, long distance and international calls.
Click on Extensions > Add SIP Extension
Extension Number: 101
Display Name: John Doe
Outbound CID: “John Doe” <0000000000> (include the “” and <>)
Secret: The SIP password for the SIP phone that John Doe is using
Click “Submit”. You how have an extension that users can reach by dialing 101. You should try to get your SIP device to register to your FreePBX server now using the extension number as the username and the secret as the password.
Click Inbound Routes > Add Incoming Route
DID Number: A phone number from your Numbers page (MUST use 11-digits: 17323395100)
Set Destination: Select the Extensions radio button and select John Doe <101>
Click “Submit”. You have now created an inbound route that will send all incoming calls to your phone number to John Doe’s phone.
Repeat steps 19-29 for each phone number to user mapping you would like to define.
Restart Asterisk
Test incoming and outgoing calls from John Doe’s phone.