TECHNICAL PREVIEWLive PSTN calling and payments are not enabledDevelopment status

DEVELOPERS / ONE OPEN CALLING PLATFORM

Build the calling
capability.

AI Dial-Tone's end-to-end design spans discovery, human permission, optional managed x402 admission, operator-controlled routing, real-time conversation, carrier-side hangup, and outcome evidence. This GitHub source is the development implementation of that product, not a separate BYOT-only offering. Version 0.0.1 provides runnable local Peer Lab media and experimental integrations, but no live public calling API.

BRING YOUR OWN TELECOMS / EXPERIMENTAL

Your Asterisk.
Your calling routes.

BYOT is for operators with an existing, isolated Asterisk/FreePBX environment and their own restricted PJSIP endpoint. AI Dial-Tone supplies an optional Python ARI/media adapter; it does not supply a SIP trunk or require managed x402 payments.

01 / PROVIDE

Use your own PBX

Prepare an isolated Asterisk system and a dedicated, restricted outbound PJSIP endpoint. Keep carrier credentials inside your telecoms infrastructure.

02 / CONFIGURE

Enable local ARI

Configure loopback-only ARI access and set the connector's ARI, fixed-trunk, operator, and separate agent credentials.

03 / VERIFY

Run the lab checks

Install Python 3.11+ and aiohttp, run the BYOT tests and python -m byot.server --check-config. Calling is disabled by default.

04 / CONTROL

Authorise a bounded call

An operator issues a one-use destination-and-duration mandate. The agent joins using its own bearer and streams audio, without seeing ARI or SIP secrets.

Your telecoms, your authority. The human permits the task; the PBX or telco independently controls whether the call can be placed, which outbound route is permitted, and when it ends. The agent cannot change routes or override a refusal. The operator owns hard timeouts and recovery after a failure; AI Dial-Tone does not create or remove PBX routes for individual calls.

0.0.1 limitation: the adapter has passed simulated ARI/UDP tests, but not a live Asterisk, SIP trunk, or PSTN interoperability test. Do not use it on a production or shared PBX.

THE 0.0.1 CONTRACT

Two authenticated bots.
One controlled session.

No public PSTN calls, no online agent account, no x402 charge and no hosted MCP. Only the local experiment below is runnable today.

GET/v1/capabilitiesLocal capability discovery
POST/v1/mandatesOperator-issued test authority
POST/v1/sessionsStart one bounded local session
POST/v1/sessions/{sid}/ticketsSingle-use peer admission
WS/v1/sessions/{sid}/connectActual PCM16 audio and attributed signals
POST/v1/sessions/{sid}/stopIndependent operator stop