The gateway can now hear. Verified end to end against the Asterisk lab:
speech classifies as live_human, hold music as music, the speech→music
transition tracks the dialplan, and DTMF reaches a real IVR (Asterisk logged
"caller pressed 1 -> accounts" and branched). RTP stats show 0% packet loss.
PJSUAEngine implements the existing SIPEngine interface, so the gateway,
call manager and hold-slayer service are unchanged. It is selected with
SIP_ENGINE=pjsua2; the default stays "sippy" while this is proven, and the
mock remains opt-in as before.
Why a new engine rather than fixing the old path: PJSUA2 exposes no
standalone RTP media object, so it will not surface media for a dialog it
does not own. Owning the dialog is the price of owning the media. Sippy keeps
the SBC roles it is good at — device registration, routing, leg bridging —
and SippyEngine remains fully functional for signalling; it simply cannot
carry media, which its media branch now says plainly instead of calling a
method that could never work.
The safety invariants are untouched. This engine is reachable only through
gateway.make_call, which refuses emergency numbers and enforces the
concurrency cap before any SIP action. No new dial path was introduced.
MediaPipeline.add_remote_stream(host, port) is replaced by
attach_call_media(stream_id, audio_media), called from onCallMediaState —
the one place PJSUA2 hands out RTP-backed media. Taps requested before media
comes up are attached when it does, so the classifier never misses the start
of a call.
Three crash/lifetime bugs found by running against the real bindings, none of
which any unit test would have caught:
- pj.Call and pj.Account objects finalised after libDestroy() abort the
process on a native assertion, exactly as media ports do. Both are now
dropped and collected before the pipeline destroys the endpoint.
- PJSUA2 keeps delivering callbacks during interpreter teardown, when module
globals may already be cleared. The callbacks alias what they need locally
and swallow everything: a raise there escapes into C++ and takes the worker
thread with it.
- hangup() only queues the BYE, so shutdown deleted the account with a call
still active and left the far end on an unclosed dialog. stop() now waits
briefly for the teardown to complete.
Threading follows the established rule: PJSUA2 worker threads reach the loop
only through _post_from_pj → run_coroutine_threadsafe, and any thread PJSUA2
did not create registers itself before touching a PJSUA2 object.
Co-Authored-By: Claude Opus 5 (1M context) <noreply@anthropic.com>
Alembic replaces create_all as the schema authority: async env.py
against Base.metadata (CLI and in-app entry paths share it via
config.attributes["connection"]), an autogenerated baseline of the
create_all-era schema, and init_db now runs upgrade head — stamping
the baseline first on a pre-Alembic database so existing deployments
adopt cleanly. create_all remains for tests only.
Calls are durable from the start: CallManager gains an
on_call_created hook (wired to persist_call_on_create) that inserts
an in_progress CallRecord the moment a call is created;
persist_call_on_end finalizes that same row. A SIGKILL mid-call now
leaves an in_progress row instead of erasing the call from history
(verified live against the dev database).
One transcript representation: ActiveCall.transcript_chunks holds
TranscriptEntry (t_offset_ms, speaker, text) — add_transcript stamps
real offsets from connect time, receptionist passes speaker instead
of encoding it into "caller: ..." strings, persisted chunks carry
real seek offsets, and the dead CallRecord.transcript Text column is
dropped by migration. Device.is_online migrates String → Boolean
(with a USING cast for existing rows).
Model de-triplication: CallResponse/CallStatusResponse build via
from_call classmethods (one ActiveCall→response mapping);
DeviceStatus deleted — can_receive_call is a computed field on
Device and the list endpoint returns the domain model; all row↔dict
and row↔domain mapping now lives in call_persistence.py
(record_summary/record_detail/chunk_to_dict + device row functions).
New tests/test_data_layer.py: upgrade-head-matches-models,
pre-Alembic adoption, durable in_progress rows, end-without-create
fallback, transcript offsets, consolidated response models.
Co-Authored-By: Claude Fable 5 <noreply@anthropic.com>
Engine mode is now explicit: USE_MOCK_SIP=true is the only way to get
the mock engine; an unconfigured trunk fails startup with guidance
instead of silently degrading. Root-caused why the engine always ran
mock: nested pydantic-settings never read .env (no env_file on the
sub-settings classes) — all 8 now declare it.
/health stops lying: reports engine mode (sippy|mock), a live DB
SELECT 1, trunk registration state with reason, and TTS/STT
availability from their last real request; "healthy" now requires
ready + db + sippy + registered trunk.
Error policy: leaf services (tts/transcription/llm_client) raise and
track availability; call-loop callers catch, publish EventType.ERROR
naming the failed service, and apply an explicit fallback. Persistence
writes get one bounded 3x exponential retry, then an ERROR log — no
more silent data loss.
Event bus: a full subscriber queue drops its oldest event (counted)
instead of silently evicting the subscription; subscribe(replay_last=N)
delivers the advertised history replay, used by /ws/events (25).
Receptionist correctness: a matched TAKE_MESSAGE rule beats the LLM;
voicemail polls for early hangup and stops/transcribes/hangs up in
finally; RecordingSession finally keeps its leg_ids so taps detach.
Dead code removed: models/contact.py + Contact table, dtmf_buffer,
transcribe_stream stub, SMS stub in notification.py.
Co-Authored-By: Claude Fable 5 <noreply@anthropic.com>
The gateway was the composition root, device registry, inbound-call
policy, and call-operations service in one class, with core↔services
circular imports papered over by function-local imports, wiring done
by assigning private attributes, and MCP tools duplicating REST query
logic against their own sessions.
Composition:
- main.py's lifespan now builds every service and wires them by
constructor/registration. gateway.from_config() is gone; core/ no
longer imports services/ anywhere — the cycle is dead.
- Inbound-call policy moved to ReceptionistService.on_inbound_call
(routing evaluation, reject/answer, screening dispatch); wired as
the engine's on_incoming_call by the lifespan. Receptionist deps
(tts/transcription/recording/routing) are constructor-injected —
no more gateway._tts reach-through or importing hold_slayer's
private _get_llm (now services.llm_client.get_llm, shared).
- Hold-slayer launch goes through a mode-handler registry
(register_mode_handler); the gateway no longer knows the service's
type. CallManager takes on_call_ended in its constructor.
- build_sip_engine() is a pure function taking explicit callbacks.
- api/routing.py uses the routing service from app.state via a
proper dependency instead of gateway._routing.
Shared data layer:
- db.session_scope() is the one session convention (get_db wraps it).
- services/call_persistence.py gains the query/write functions and
the single StoredCallFlow→CallFlow mapper; api/call_flows.py,
api/call_history.py, and the six DB-touching MCP tools are thin
wrappers over them — the two surfaces can't drift.
- legs_for_call() replaces the three private _call_legs scans
(gateway transfer/hangup, REST dtmf, MCP dtmf).
7 new tests (mode-handler launch, on_call_ended hook, receptionist
inbound answer/reject, call-flow CRUD round-trip and history routes
against real SQLite through the shared layer). aiosqlite added to dev
deps for that.
Co-Authored-By: Claude Fable 5 <noreply@anthropic.com>
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.
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
