SippyEngine had never successfully placed a call or registered a trunk.
Every failure was masked by broad exception handlers that logged and marked
the leg terminated, so the gateway reported "ringing" and then ended the
call rather than surfacing the fault. None of it was visible to the test
suite, which runs exclusively on MockSIPEngine.
Found by pointing the gateway at a local Asterisk instance (tests/lab) —
each fix uncovered the next.
1. Trunk registration used kwargs the installed sippy (2.3.0) does not
accept (auth_name/auth_password → user/passw), called register()
instead of doregister(), and passed aor/contact as strings where
SipRegistrationAgent calls .getCopy() and mutates .username/.port,
so SipURL objects are required.
It also posted registered=True at *send* time. Registration is
asynchronous, so a rejected REGISTER would still have reported success
— and /health treats a registered trunk as a condition for "healthy".
Now wired to sippy's rok_cb/rfail_cb, so the rejection status line
(typically a bad trunk password) reaches the operator.
The Contact also fell back to loopback when the SIP bind is 0.0.0.0;
a wildcard address is not somewhere a trunk can send an INVITE.
2. The INVITE passed SDP as a `body` kwarg. CCEventTry takes no such
argument: UacStateIdle unpacks exactly six fields from the data tuple
and expects the SDP as a MsgBody in position four. callingID/calledID
are bare usernames — sippy builds the URIs itself from nh_address.
3. _sip_logger was absent from the global config. SipTransactionManager
dereferences it on every message, so the first SIP packet in either
direction raised KeyError inside the ED thread.
4. SippyCallController was not callable. Sippy invokes event_cb(event, ua)
with CCEvent objects; the class only exposed on_* methods that nothing
called. Added __call__ to dispatch CCEventRing/Connect/Disconnect/Fail
to the existing handlers, guarding the body because an exception
escaping into the ED dispatcher would hang the leg silently.
5. The UA was constructed without credentials, so sippy could not answer
the 401/407 challenge that any authenticating trunk sends. Every
outbound call died on the challenge.
Verified end to end against Asterisk 22.10.1: 180 Ringing → Connected →
23s of audio → clean teardown, with the dialplan executing and audio
playing in real time.
Not fixed here, and still blocking media: MediaPipeline.create_tap is a
stub that logs success and returns a tap nothing ever feeds, so the
classifier receives no audio on a live call.
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
