FreePBX AI Receptionist: Replace Your IVR Menu with an AI Agent

A FreePBX AI receptionist answers your main number, asks the caller how it can help, and then does what a good front-desk person would: answers simple questions, puts the caller through to the right extension or queue, or takes a message when no one is free. It replaces the “press 1 for sales, press 2 for support” menu with a conversation, and it runs on the FreePBX system you already have.

This guide focuses on the receptionist role: what it should do, how to design the call flow, how it connects to FreePBX, and how transfers to your extensions work. If you want a broader setup walkthrough first, see our post on VEXYL as a FreePBX AI agent.

What a FreePBX AI receptionist does

Think about what your best receptionist handles in a day. Most of it fits into a short list:

• Greets and identifies the need: “Good morning, Sharma and Associates. How can I help?”
• Routes by name, department or reason: “Can I speak to Priya in accounts?” or “I have a question about my invoice.”
• Answers common questions: office hours, address, directions, which documents to bring.
• Takes messages when the person is busy or it is after hours, and sends them to the right inbox or system.
• Screens calls: asks for the caller’s name and reason before putting them through.
• Speaks the caller’s language, switching between English and Hindi, Tamil or Malayalam as needed.

A FreePBX AI assistant can do all of these. What it should not do is make decisions that need authority, such as approving discounts or handling complaints. Those go to a person, quickly.

AI receptionist vs a FreePBX IVR menu

For a deeper comparison of menus and conversational systems, read our guide to AI IVR systems.

Designing the receptionist call flow

Before touching the dialplan, write the flow on paper. It will save hours of rework.

1. Build the routing table

List every destination the receptionist can transfer to: extensions, ring groups and queues, with the names and reasons that should lead there. For example, “accounts, billing, invoice, payment” all go to the accounts ring group. Include alternative names people use, nicknames and common misspellings of staff names.

2. Write the greeting

Short and open. Name the organisation, say it is an automated assistant, and ask how it can help. Do not list departments; that is what you are moving away from.

3. Confirm before transferring

“I’ll connect you to the accounts team. Is that right?” A one-second confirmation prevents most wrong transfers.

4. Decide what happens when nobody answers

If the transfer target is busy or does not pick up, the call should come back to a sensible place: a queue, voicemail or the AI taking a message. Plan this in FreePBX the same way you would for any inbound route.

5. Write the after-hours behaviour

Outside office hours, the receptionist can answer FAQs, take a message with name, number and reason, and tell the caller when they will hear back. FreePBX time conditions can send calls to a different bot or the same bot with after-hours instructions.

6. Add the information it should know

Hours, holidays, address, directions, parking, services and prices you are happy to share. Keep it short and accurate, and assign someone to update it.

How the AI receptionist connects to FreePBX

FreePBX runs on Asterisk, so the cleanest connection is Asterisk’s AudioSocket application, which streams call audio to a voice gateway over TCP. The steps, at a high level:

1. Install the voice gateway on the FreePBX server or a nearby one.
2. Add a custom dialplan context in extensions_custom.conf that answers the call, creates a session ID, posts the caller’s details (number, name, channel, which bot and language) to the gateway, and hands the audio over with AudioSocket().
3. Create a Custom Destination in FreePBX that points at that context.
4. Point an inbound route, a time condition or an IVR option at the Custom Destination.
5. Configure transfers so the AI can send callers back into FreePBX extensions, ring groups or queues.

The core of the dialplan looks like this:

[voice-assistant]

exten => _X.,1,Answer()

same => n,Set(SESSION_UUID=${TRIM(${SHELL(uuidgen)})})

same => n,AudioSocket(${SESSION_UUID},127.0.0.1:8080)

same => n,Hangup()

In production you add a step before AudioSocket() that posts the caller ID, channel name, bot and language to the gateway. The Asterisk AudioSocket documentation has the full dialplan, firewall settings and per-number bot selection.

If you prefer SIP, FreePBX can also reach the gateway over a pjsip trunk, and the SIP trunk guide includes the FreePBX GUI settings.

How transfers to extensions work

Transfers are what make an AI agent a receptionist rather than an answering machine. With AudioSocket, when the AI decides to transfer, the gateway sends an HTTP request with the call’s Asterisk channel name to a transfer endpoint you provide. That endpoint moves the channel, for example with an AMI Redirect into a queue or extension context, and replies with success. This is why the channel name must be posted with the caller details at the start of the call: without it, the transfer cannot find the call.

For multiple destinations, the usual pattern is to have the AI choose a destination from your routing table and pass it along, so the transfer script can redirect to the right extension, ring group or queue.

Testing checklist before going live

• Call from a mobile in a noisy place and ask for each department by different names.
• Ask for staff by first name, full name and nickname.
• Interrupt the greeting to check barge-in works.
• Test transfers to busy and unanswered extensions.
• Test after-hours routing and message taking.
• Ask something the receptionist does not know and confirm it offers a transfer instead of guessing.
• Test in every language your callers use.

How VEXYL fits

The VEXYL Voice Gateway is built for this setup. It connects to FreePBX and Asterisk over AudioSocket (or SIP), serves different bots, languages and voices per number from one install, and supports transfers through your transfer endpoint on AudioSocket calls or SIP REFER on SIP calls. It handles barge-in, echo and line noise, and plays a short filler sound while the AI thinks so callers do not hear dead air.

You pick the speech providers for your callers’ languages, including Sarvam, Google, Deepgram and the self-hosted VEXYL-STT, and the LLM behind the bot. Because it runs alongside FreePBX on your own server, call audio and transcripts can stay inside your network, as described in our guide to self-hosted AI voice agents. The free Developer edition supports 4 concurrent calls, which is enough to run a receptionist pilot on a real number.

FAQ

Does FreePBX have a built-in AI receptionist?

FreePBX provides IVRs, ring groups, queues and time conditions, but an AI receptionist typically comes from an external voice gateway connected through Asterisk AudioSocket or a SIP trunk.

Can the AI receptionist transfer calls to FreePBX extensions?

Yes. On AudioSocket calls, the gateway calls a transfer endpoint that moves the Asterisk channel to an extension, ring group or queue. On SIP calls, it can use SIP REFER.

Can I keep my existing FreePBX IVR?

Yes. You can add the AI receptionist as one IVR option, use it only after hours, or put it on a single inbound route while the rest stays unchanged.

What does a FreePBX AI voice agent need to run?

A FreePBX or Asterisk system with AudioSocket available, the voice gateway on the same or a nearby server, and API keys for the speech and AI providers you choose, unless you self-host them.

How many calls can it handle at once?

That depends on your licence, server and provider limits. The free Developer edition allows 4 concurrent calls; Professional allows 25.

Ready to give your FreePBX an AI receptionist? Download the free Developer edition, or talk to the VEXYL team for help with setup.