Configure FreePBX 17 — registration trunk
Set up a chan_pjsip registration trunk on FreePBX 17, including the auth credentials and inbound context.
Last updated March 27, 2025
This guide sets up a SIP trunk on FreePBX 17 using PJSIP, which handles NAT better than the older chan_sip driver and supports multiple registrations. If you would rather authenticate by IP address, use the IP trunk guide instead.
Before you start
- Download and install FreePBX 17, and configure its administration and firewall settings.
- Create a SIP trunk in the Wiretap Portal and assign DIDs to it before you start here.
Check your SIP settings
- 1Go to Settings → Asterisk SIP Settings.
- 2Enter your external and local network addresses in both General SIP Settings and PJSIP Settings.
- 3Click Submit, then Apply Config.
Create the trunk
- 1Go to Connectivity → Trunks and select Add SIP (chan_pjsip) Trunk.
- 2On the General tab, name the trunk and set the outbound caller ID.
- 3Open the pjsip Settings tab and set your dialed number manipulation rules. The Dial Patterns Wizard will generate sensible US patterns for you.
- 4Still on pjsip Settings → General, fill in the connection details below.
Set a caller ID somewhere
If you do not set an outbound CID on the trunk, you have to set one on every applicable extension. With neither in place, your calls reach us without a valid caller ID.
- Auth Username
- The User ID generated when you created the trunk in the Wiretap Portal.
- Secret
- The password you generated when you created the trunk in the portal.
- SIP Server
srv.globalsbc.core-trunk.com- SIP Server Port
- Leave blank when using the SRV record. It defaults to port
5060. - Context
from-pstn-toheader
Apply the changes
- 1Select Apply Config.
- 2Add your extensions and inbound and outbound routes.
Still stuck?
Our help desk answers on 816 WIRETAP, or open a ticket. Including the right details gets you a real answer on the first reply — see submitting support tickets.
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