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FreePBX Caller ID Setup

2026-03-15

OVERVIEW

FreePBX provides a web-based GUI for managing Asterisk. This guide walks you through adding SpoofGlobal as a SIP trunk and configuring outbound caller ID spoofing — all through the FreePBX interface, no command line needed.

STEP 1: ADD SIP TRUNK

  1. Go to Connectivity > Trunks in FreePBX
  2. Click Add Trunk > Add SIP Trunk
  3. Set the trunk name (e.g., "SpoofGlobal")
  4. Under SIP Settings > Outgoing, set the trunk peer details with SpoofGlobal's server IP
  5. For SIP auth: add your username and secret from the bot
  6. For IP2IP: set type=peer and insecure=invite,port — no credentials needed
  7. Save and apply config

STEP 2: CREATE OUTBOUND ROUTE

  1. Go to Connectivity > Outbound Routes
  2. Click Add Outbound Route
  3. Set a name (e.g., "International")
  4. Under Dial Patterns, add patterns for the numbers you want to route (e.g., 1NXXNXXXXXX for US)
  5. Under Trunk Sequence, select the SpoofGlobal trunk
  6. Save and apply config

STEP 3: SET CALLER ID

There are multiple ways to control caller ID in FreePBX:

The priority order is: Route CID overrides Extension CID, which overrides Trunk CID. If none is set, the default from SpoofGlobal's bot settings is used.

STEP 4: TEST

Pick up a phone connected to your FreePBX and dial an external number. Check that the call connects and the correct caller ID appears on the recipient's phone.

If calls fail, check Reports > Asterisk Logfiles for errors. Common issues include incorrect trunk credentials, missing outbound routes, or insufficient balance.

Can I use FreePBX with IP2IP?+

Yes. Set up the trunk without credentials and use IP2IP mode in the SpoofGlobal bot. Your FreePBX server's IP must be static and whitelisted.

Does this work with FreePBX 16?+

Yes. SpoofGlobal works with all FreePBX versions that support SIP trunking, including FreePBX 14, 15, and 16.

Can different extensions have different caller IDs?+

Yes. Set the Outbound CID field on each extension individually. You can also use outbound route CID to override all extensions for specific routes.

How do I troubleshoot failed calls?+

Check the Asterisk full log at Reports > Asterisk Logfiles. Look for SIP 4xx errors. Verify your balance, route, and trunk settings. Contact SpoofGlobal support if needed.

READY TO START?

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