Call, don't configure.
A single interface for agents. No SIP accounts, softphones, dialplans, or carrier credentials in your model's context.
AI DIAL-TONE / AUTONOMOUS AGENTS
Give your AI
Dial-Tone
An end-to-end, open-source calling platform for autonomous AI agents. From human authorisation and optional x402 payment to provider-controlled routing, real conversation, hangup, and a grounded outcome. Managed calling or your own telecoms. Currently in development.
Open source · Model-neutral · Managed or bring your own telecoms · 0.0.1 preview
AI DIAL-TONE / CALL SESSION
AGENT REQUEST 01 / 04
“Call the shop and ask if the part is available today.”
01
AuthorisePermissions & limits
✓02
Pay when requiredx402 for managed calling
○03
Connect & converseAttempt a real call, then confirm media
—04
ReportEvents, usage & outcome
—NO SIP CONFIGURATION IN YOUR AGENT
TARGET INTERFACE (NOT YET LIVE)MANAGED x402 / SELF-HOSTED TELECOMSCONVERSATION / CONTROL / EVIDENCE
WHY THIS EXISTS
A real phone call.
On an agent's terms.
Give the task to an agent. It should obtain permission, connect to the telephone network, hold a conversation and return a grounded result. The telecoms complexity stays out of the way.
Built for conversation.
Real-time, bidirectional audio is the aim. Listen, speak, respond to interruption, and know when a person has answered.
Trust the event, not the guess.
Sequenced events, explicit call states, and attributed results. An agent's claim is never confused with network-confirmed reality.
Bounded by design.
Destination permissions, maximum durations, identity-bound credentials, and independently verified termination are essential.
THE CALLING SYSTEM
One agent interface.
Two telecom routes.
Design target: a human authorises a bounded task, the agent satisfies any managed x402 payment requirement, and the chosen provider executes the call through controlled FreePBX/PJSIP call policy. Operators may instead supply their own Asterisk infrastructure without managed payments.
Discover
Read capability flags; request human permission for one destination. No SIP credentials in model prompts.
Authorise and pay
Managed calling aims to use x402; BYOT needs no managed payment. Neither payment nor speech authorises an arbitrary route.
Connect and converse
Use the selected FreePBX/PJSIP connection to establish a bounded call and carry two-way speech. Readiness and routing checks reduce avoidable failures, but do not guarantee an answer.
Report accurately
Separate network state, what was heard and what the destination actually confirmed. Never manufacture success.
THE DEVELOPER EXPERIENCE
Four steps.
One conversation.
What the finished service will feel like to an autonomous agent. The laboratory currently simulates the calling boundary.
01
Describe the task
Who to call, why, which agent is acting, and the maximum duration.
02
Receive a call ID
Use idempotency keys so retries don't create multiple calls.
03
Stream the conversation
Audio flows separately from the control API; events tell the agent what changed.
04
Return the evidence
Track status, usage, and what the agent observed, without overstating certainty.
DESIGNED FOR REAL SOFTWARE
An API you can reason about.
Clear input, a stable call identifier, a replayable event stream, and honest failure states.
Explore the APIrequest.json
{
"destination": "1002",
"task": "Ask the test agent whether an item is available",
"agent_id": "stock-checker",
"max_seconds": 30
}LAB DESTINATION ONLY · NOT A REAL PHONE NUMBER
EXPERIENCE BEHIND THE PROJECT
Not new to
FreePBX.
AI Dial-Tone is developed by Kieran Knowles-Byrne (@kierknoby), an open-source FreePBX community-module developer and Head of Product at FreePBX UK. It draws on years of operating real-world Asterisk, PJSIP and hosted telecommunications.
20tele & FreePBX UK
20tele has spent more than a decade working with FreePBX and Asterisk. FreePBX UK holds Sangoma's Official FreePBX partner designation for the UK. These are credentials of the people and associated businesses, not an endorsement of this experimental project.
FreePBX UK ↗20tele ↗Community modules
Registration Watch, Repeat Caller and Concurrency Count are real, publicly inspectable FreePBX community projects. The same priority runs through this work: useful diagnostics, evidence, operational safeguards and maintainable source.
Registration Watch ↗Repeat Caller ↗Concurrency Count ↗FOR EVERY CAPABLE MODEL
Your model.
Your workflow.
Our connection.
One intended agent-facing capability, from approved task to verified call closure and outcome. Developers can integrate a model or supply their own telecoms. The 0.0.1 repository provides local experiments; no hosted calling service is available yet.
For builders01
REST
→Planned hosted commands and replayable events; currently only local Peer Lab HTTP control.
02
MCP
→Planned; there is no MCP tool server in the 0.0.1 release.
03
REAL-TIME MEDIA
→Implemented for localhost Peer Lab PCM; no telephone media gateway yet.
04
DIAGNOSTICS
→Local event evidence works; real SIP/call-state verification is future work.
THIS IS A TECHNICAL PREVIEW
See how it works.
No phone call required.
Run five deterministic scenarios in your browser, inspect the events, and explore what an agent would receive.
CURRENT REALITY
Ambitious product.
Precise claims.
The control plane, event journal, credentials, simulation, and audio protocols have local test coverage. Actual PSTN calling, independent carrier cutoff, payment settlement, and hosted speech-model conversation still need live verification.
See exactly what's been tested →