Your numbers and carriers
Trunks and numbers stay on the PBX. Point a chosen line or internal extension at Chat411. Other routes keep working as they do today.
An AI receptionist for the phone systems you already support.
Early release · Answer-and-transfer is live
Front desks miss calls and after-hours lines roll to voicemail. Most AI voice products want the whole phone system moved onto their platform. Chat411 connects to Asterisk and FreePBX instead, so numbers, carriers, and routing stay where they are. If you are an MSP, the client stays with you.
Chat411 can face the PSTN or stay on-net. The local path uses a local model with media that can be encrypted. The cloud realtime path streams caller audio to a provider, xAI by default.
Add a receptionist, not a new platform.
Your dial plan
Transfers go to any extension reachable from your internal dial plan. The agent hands the caller over and leaves the call.
Your client relationship
You deploy and support it on systems you already manage. Your client doesn’t move to another phone platform.
What your PBX needs.
Chat411 is a .NET service on your network. This is what it expects from the phone system.
| Requirement | Detail |
|---|---|
| PBX version | FreePBX 16 or 17, or Asterisk 18 and later |
| Call control | Asterisk REST Interface (ARI) enabled, with a user for Chat411 |
| Media | WebSocket external media, carrying 24 kHz audio in the format Asterisk already uses |
| Dial plan | A Stasis entry point and a route from your chosen line or internal extension |
| Network | The PBX reaches Chat411 for media, and Chat411 reaches ARI. Keep both on trusted networks, not the public internet. |
| Local path | A local conversation model. Media can be encrypted. No cloud model provider or PSTN connection is required. |
| Cloud realtime path | Outbound access to a cloud realtime voice provider, xAI by default. Caller audio goes to that provider. |
Both voice paths are supported now. The local path can keep conversation audio on your network. The cloud realtime path sends it to a provider. Call control stays on your PBX in either case.
How a call moves from ring to handoff is on the PalmConnect page.
Start with one line or extension.
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Pick one line
Choose an internal extension, after-hours line, or overflow number. Decide between the local and cloud realtime paths before connecting calls.
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Connect Chat411
Enable ARI and external media, add the Stasis entry point, and route the chosen line or extension to it.
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Test transfers
Call in, ask for people by extension, and confirm each transfer lands where your dial plan says it should.
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Watch, then expand
Follow live transcripts in the call monitor, then add more numbers or sites.
Someone still picks up.
PalmConnect Phone is a native desktop SIP softphone for the person taking the call. Windows-first · Early 0.1. Register an extension, place or answer a call, and switch on RNNoise on the device. This denoising is separate from either Chat411 conversation path. View PalmConnect Phone
Where it is going
Lookups in your clients’ systems are groundwork built, with a person approving anything that changes a record. A configured front desk, saved call records, and outbound calls are planned. More on how we build AI agents, and the full PalmConnect roadmap.
Questions from PBX admins and MSPs
Does Chat411 replace the PBX?
No. It runs next to Asterisk or FreePBX as a separate service. Your numbers, carriers, trunks, and routing stay on the PBX.
Which phone systems does it work with?
FreePBX 16 or 17, or Asterisk 18 and later, with the Asterisk REST Interface and WebSocket external media enabled. The PalmConnect Phone softphone also works with other WebRTC-capable SIP PBXs such as FreeSWITCH and 3CX.
Where does caller audio go?
You choose one of two supported paths. The local path uses a local model, media can be encrypted, and PSTN is not required; audio can stay on your network. The cloud realtime path streams caller audio to a provider, xAI by default. Call control stays on your PBX in both paths.
What does the caller hear?
A greeting as soon as the call connects, with no menu to press through. The caller says what they need, and the agent answers or transfers them when they ask for a person.
Can it do more than answer and transfer?
Not yet. Answer and transfer are live today. Lookups in your systems are groundwork built, and a configured front desk, saved call records, and outbound calls are planned. See the roadmap.
Hear it answer a line you support.
We will walk through Chat411 against a real Asterisk or FreePBX setup and what a first route would look like.