2026-03-15
Asterisk is the most widely used open-source PBX in the world. Combined with SpoofGlobal's SIP trunking, it gives you complete control over outbound caller ID for every call. You can set different caller IDs per extension, per route, or dynamically per call using the Asterisk dialplan.
There are two ways to connect your Asterisk server to SpoofGlobal:
If your Asterisk uses chan_sip, add a trunk in sip.conf:
type=peerhost to SpoofGlobal's server IPusername and secret from the botinsecure=invite for IP2IP trunksfromdomain to the server IPFor PJSIP, create an endpoint, AOR, and optionally an auth section. For IP2IP mode, you only need the endpoint and AOR — no auth section required.
The simplest way to set caller ID per-call is in extensions.conf:
Set(CALLERID(num)=18005551234) before the Dial commandSet(CALLERID(name)=My Company) for the display nameFor dynamic spoofing, you can read the desired caller ID from a database, a variable, or user input via an AGI script.
Create a dialplan context that routes calls through the SpoofGlobal trunk. Match the destination pattern, set your desired caller ID, and dial through the trunk.
Use the Asterisk CLI (asterisk -rvvv) to watch SIP messages and verify your caller ID is being sent correctly. Check that the From header contains the number you set.
Yes. FreePBX is built on Asterisk. Add SpoofGlobal as a SIP trunk in the FreePBX GUI and set outbound caller ID in your outbound routes.
Yes. Use the Asterisk dialplan to set CALLERID(num) dynamically based on the calling extension, the destination, or any other variable.
Any version that supports SIP or PJSIP trunking. Asterisk 13+ is recommended for best compatibility.
Only for IP2IP mode. With SIP registration, your Asterisk can be on any IP — it authenticates with username and password.