Jev Call Router

Clone the repository, run bun install, copy .env.example to .env and fill in values; .env is gitignored. Use bun start for the service or bun run demo for the demo runner.

TYPESAFE_API_KEY is required. STT_API_KEY, STT_BASE_URL, and STT_MODEL are required for speech-to-text. ARI_URL, ARI_USERNAME, and ARI_PASSWORD are required for bun start. Other variables such as TYPESAFE_DEFAULT_MODEL, STT_LANGUAGE, ARI_APP, ANTHROPIC_API_KEY, WEBHOOK_URL, WEBHOOK_SECRET, and PORT are optional.

It supports Asterisk 16+ with ARI enabled, or FreePBX 16/17, which runs Asterisk underneath. The core router, Jev provider, and types do not depend on Asterisk; a PBX adapter only has to implement the CallSession interface.

Each decision is one choice question whose options are your described routes plus need-more-information and no-matching-route. The router validates that the route is known and confidence is at least minConfidence; low confidence asks again up to maxAttempts, while invalid, failure, or timeout uses fallbackRoute.

Caller audio is sent to a speech-to-text service over an OpenAI-style /audio/transcriptions endpoint. An optional Claude interpreter summarizes the transcript and removes personal details. Jev receives the transcript and route options, and an optional webhook can receive routing outcomes signed with HMAC-SHA256.

It is a TypeScript service and library that requires Bun 1.3+. It uses the official @typesafe-ai/sdk systemOne API for Jev decisions.

Deterministic rules run first via routeDirectly(call) for VIP lists, business hours, per-number opt-in, or dialplan flags. When routeDirectly returns a route, Jev is not called and the deterministic route is used.