Three thread domains were mutating shared dicts with no locks: Sippy's
ED thread wrote _legs/_registered_devices directly from SIP handlers,
the asyncio loop wrote them from make_call/hangup, and
run_in_executor(None, ...) had default-pool threads driving sippy UA
objects. AudioTap.feed() pushed into an asyncio.Queue (not
thread-safe) from the PJSUA2 thread.
New ownership rule, enforced structurally:
- The asyncio loop owns all app-visible state; the only mutator is the
new _on_engine_event funnel. Sippy handlers extract plain strings on
the ED thread and post via run_coroutine_threadsafe.
- The ED thread owns sippy objects plus _ed_ua_to_leg/_ed_leg_to_ua;
loop-side commands (INVITE/BYE/DTMF/trunk register) hop over via
ED2.callFromThread. UA references no longer live on SipCallLeg.
- AudioTap captures its loop and feed() hops via call_soon_threadsafe.
- Fix ED import: installed sippy 2.x exposes ED2, not ED — the old
import could never start the event loop.
Also:
- Wire the never-connected on_leg_state_change callback: outbound
ringing/connected/terminated now reaches CallManager; a call ends
when its last leg terminates (transfers keep it alive). Adds
CallManager.unmap_leg/legs_for_call.
- AudioClassifier.classify(): async entry that runs the FFT work in
asyncio.to_thread and updates history on the loop — all four
hold_slayer call sites now route through it, fixing both the
loop-blocking and the 2-of-4 history gap. DTMF Goertzel loop
replaced by the equivalent vectorized DFT-bin power.
- Task hygiene: gateway.spawn() tracks hold-slayer/receptionist tasks
and stop() cancels them; recording safety-timeout task is retained
and cancelled on stop_recording; engine tracks incoming-call
dispatch tasks.
10 new tests: funnel events from a foreign thread, auto-answer
fallback, AudioTap cross-thread feed, classifier history, leg-state →
call status (including no stomping of ON_HOLD), stop() cancellation.
Co-Authored-By: Claude Fable 5 <noreply@anthropic.com>
The MCP server was created but never mounted — no client could reach
it. Mount it at /mcp/ over streamable HTTP with a combined lifespan,
resolving the gateway lazily so mounting happens at app construction.
Security and safety for the agent surface:
- One static API_TOKEN (SecretStr) enforced across REST (dependency),
WebSocket (query param/header before accept), and MCP
(StaticTokenVerifier). Startup refuses tokenless non-loopback binds.
- Emergency numbers (911/9911/112) always refused on make_call, plus a
MAX_CONCURRENT_CALLS cap; ValueError surfaces as 400/ToolError.
- Safe defaults: debug off, no credential in default DATABASE_URL,
SIP/LLM/TTS secrets as SecretStr.
Cleanups:
- Delete broken learn_call_flow tool (wrong ctor args, nonexistent
method) and the never-fed CallAnalytics service; keep
call_flow_learner for proper wiring later.
- Trim dial_plan to what is actually used (emergency guard, extension
allocation); delete the unreferenced matcher/normaliser.
- Register call_history before calls so /api/calls/history is no
longer shadowed by /api/calls/{call_id}.
- fastmcp pinned >=3.0 (http_app + StaticTokenVerifier).
New tests: MCP in-memory client (tool surface, lazy gateway, emergency
refusal, call cap) and API security (401 paths, route order, mount).
Co-Authored-By: Claude Fable 5 <noreply@anthropic.com>
Adds read-only access to persisted call records for the dashboard
and implements a client for the Rhema text-to-speech service.
- api/call_history.py: New router providing paged call lists
and detailed call records with transcript metadata.
- services/tts.py: Async client for OpenAI-compatible TTS
endpoints (Rhema/Kokoro) used for call-flow steps.
- Simplified .env.example to use localhost SPEACHES_URL
- Removed unused prod_url from SpeachesSettings config
- Added dashboard node_modules and build dirs to .gitignore
- Streamlines local development setup
Complete project scaffolding and core implementation of an AI-powered
telephony system that calls companies, navigates IVR menus, waits on
hold, and transfers to the user when a human answers.
Key components:
- FastAPI server with REST API, WebSocket, and MCP (SSE) interfaces
- SIP/VoIP call management via PJSUA2 with RTP audio streaming
- LLM-powered IVR navigation using OpenAI/Anthropic with tool calling
- Hold detection service combining audio analysis and silence detection
- Real-time STT (Whisper/Deepgram) and TTS (OpenAI/Piper) pipelines
- Call recording with per-channel and mixed audio capture
- Event bus (asyncio pub/sub) for real-time client updates
- Web dashboard with live call monitoring
- SQLite persistence via SQLAlchemy with call history and analytics
- Notification support (email, SMS, webhook, desktop)
- Docker Compose deployment with Opal VoIP and Opal Media containers
- Comprehensive test suite with unit, integration, and E2E tests
- Simplified .gitignore and full project documentation in README
