FreePBX

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Prerequisites

• FreePBX admin access
• Outbound UDP 5060 allowed on the firewall
• SIP ALG disabled on the router (mandatory)
• PBX SIP Domain (FreePBX’s public FQDN or IP)

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Step 1: create the trunk in SmartAlex

Phone Numbers, then Connect Your Carrier, then SIP Trunk, then pick FreePBX. Save, copy the credentials.

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Step 2: add the pjsip trunk in FreePBX

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Step 3: inbound route

Connectivity, then Inbound Routes, then Add. Set the DID, destination as your SmartAlex trunk (outbound leg). Alternatively if using a SmartAlex DID, skip this step.

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Step 4: outbound route (optional)

Only if FreePBX extensions should dial out through SmartAlex. Connectivity, then Outbound Routes, then add a dial pattern that strips the prefix and uses the SmartAlex trunk.

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Step 5: extensions and agent linking

Add extensions in SmartAlex Settings, then PBX. Link agent in Agent Studio.

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Step 6: test

Dial the DID. Ask for a transfer.

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Alternative: Native Extension Mode

The flow above has FreePBX register out to SmartAlex via a trunk. The inverse, SmartAlex registers as an extension on YOUR FreePBX, is often simpler when FreePBX is your source-of-truth. Provision a normal pjsip extension in Applications → Extensions, give it to SmartAlex, and we register exactly like any softphone. From there, route DIDs through your usual inbound-route + ring-group rules. See Native Extension Mode.

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FreePBX-specific notes

• SIP Debug: Admin, then Asterisk CLI. Run pjsip set logger on to trace SIP messages in real time.
• NAT handling: pjsip handles NAT well when rewrite_contact is enabled (default).
• Call recording conflicts: if FreePBX records calls at the trunk level, it will record the SmartAlex leg separately. Decide whether that’s wanted.
• Distro vs Incredible PBX vs stock FreePBX: configuration is identical across distributions.

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Next steps