Release 1.2.0a1 - #49
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AI-Generated Change: - Model: claude-sonnet-4-6 - Intent: Replace setup.py with PEP 621 pyproject.toml; add version.py for dynamic version via attr: - Impact: Package installs via modern toolchain; setup.py removed - Verified via: pip install -e . && pip show baresipy
AI-Generated Change: - Model: claude-haiku-4-5 - Intent: Add standard CI/CD workflows via shared-gh-workflows skill - Impact: Added workflow files for build, test, lint, release automation - Verified via: apply_workflows.py metadata detection
- fix readme filename mismatch (readme.md -> README.md) - point Homepage at TigreGotico/baresipy - declare json_database dependency and a test extra - require python >=3.10 to match the CI matrix - drop legacy hand-rolled workflows superseded by gh-automations - drop lichecker license test superseded by the license_check workflow - fix example control flow and note the pyjokes example dependency Co-Authored-By: Claude Fable 5 <noreply@anthropic.com>
audioop-lts backfills the audioop module pydub needs on python >= 3.13 Co-Authored-By: Claude Fable 5 <noreply@anthropic.com>
chore: packaging & repo hygiene
…ess calls BREAKING CHANGE: ResponsiveVoice is removed. The `tts` kwarg now expects an object exposing the OVOS TTS plugin interface (get_tts(sentence, wav_file)). A default engine can be resolved lazily via the optional `ovos` extra (ovos-plugin-manager + ovos-tts-plugin-phoonnx). - extract per-line parsing out of run() into _handle_output_line() - add type hints across the public API; make quit() idempotent and tolerant of an already-dead baresip process - add autostart kwarg to defer Thread.start(); block semantics now live in an overridden start() - add login_options kwarg to extend the SIP login string - support registrar-less mode: user/pwd/gateway all optional, call() accepts full sip: URIs or expands number+gateway - add headless mode (config.render_config) so baresip can run without any sound hardware (fixes #16, #17); store the configured ausrc so send_audio() stops hardcoding alsa,default - harden contacts import/export against a missing contacts file (#2) - convert_audio() gains frame_rate/channels/min_duration_s kwargs (#18) - fix ruff re-export warning in baresipy/utils Co-Authored-By: Claude Fable 5 <noreply@anthropic.com>
baresip 3.x logs DTMF as 'received SIP INFO DTMF' (SIP INFO) or 'received in-band DTMF event' (RTP telephone-event), and interpolates the errno text into the audio-source failure message Co-Authored-By: Claude Fable 5 <noreply@anthropic.com>
feat!: pluggable TTS, headless mode, registrar-less calls
Covers _handle_output_line() line parsing (ready/login/call states/DTMF/ errors), render_config() headless vs alsa, ContactList CRUD, convert_audio() resampling, and get_default_tts()/speak() fallback behaviour, all without spawning a real baresip process or touching ~/.baresipy or ~/.baresip. Co-Authored-By: Claude Fable 5 <noreply@anthropic.com>
test: functional unit suite for state machine, config, contacts, audio and tts
the automated version bump appended a VERSION_BLOCK because the file predated the expected format, corrupting it and regressing the computed version to 0.0.1; the version is set to 1.0.0a2 reflecting the major bump from the breaking TTS change plus the merges since 0.1.5 Co-Authored-By: Claude Fable 5 <noreply@anthropic.com>
fix: restore canonical version block format
Adds baresip sndfile-based rx call recording (record_rx/recording_path on BareSIP, get_rx_wav/get_rx_stream), a WavTailReader + pure-stdlib PCM resampler in baresipy/audio.py, an OVOS Microphone plugin (baresipy.ovos.BareSIPMicrophone, lazy-imported behind the ovos extra), and a registrar-capable voice_bot.py example wiring it into ovos-simple-listener. Co-Authored-By: Claude Fable 5 <noreply@anthropic.com>
feat: inbound call audio capture and OVOS microphone integration
Add a Dockerfile (python:3.12-slim + apt baresip/ffmpeg, optional INSTALL_EXTRAS build-arg) and a docker-compose.e2e.yml running two registrar-less headless BareSIP instances (callee/caller) that call each other directly over a compose network, exchange DTMF and a generated sine-wave wav, and write results to a shared bind-mounted volume. test/e2e/test_call.py drives `docker compose ... up` and asserts the call actually establishes and the callee's rx capture is non-silent, exercising baresipy's real stdout parser against an actual baresip binary. Marked `e2e` and excluded from the default run via addopts in pyproject.toml. Co-Authored-By: Claude Fable 5 <noreply@anthropic.com>
registrar-less mode now creates a local non-registering user agent (;regint=0) so direct sip: URI calls actually work; send_dtmf gained a 'keys' mode that emits real RFC 4733 telephone-events; the run loop survives the pty closing mid-read Co-Authored-By: Claude Fable 5 <noreply@anthropic.com>
feat: docker image, GHCR publishing and live e2e call test rig
Cover system dependency install, SIP account setup, direct/registrar-less calling, configuration reference, OVOS voice-bot integration, docker usage, and testing, and refresh AGENTS.md with the current module layout. Co-Authored-By: Claude Fable 5 <noreply@anthropic.com>
docs: full setup documentation
resolves the baresip executable via the optional baresip-binary package first, then falls back to the system binary on PATH Co-Authored-By: Claude Fable 5 <noreply@anthropic.com>
feat: prefer the baresip-binary bundled executable when installed
Expose a BareSIP phone over HTTP + WebSocket (baresipy-gateway console script) so calls, TTS, DTMF, and caller audio can be driven from any process, not just embedded python. fastapi/uvicorn/python-multipart are only required by the new server extra; base install is unaffected. Co-Authored-By: Claude Fable 5 <noreply@anthropic.com>
feat: HTTP/WebSocket gateway (baresipy.server)
httpx 1.0 removes BaseTransport, which starlette's TestClient still imports, breaking test collection Co-Authored-By: Claude Fable 5 <noreply@anthropic.com>
Co-Authored-By: Claude Fable 5 <noreply@anthropic.com>
fix: cap httpx below 1.0 in the test extra
Co-Authored-By: Claude Fable 5 <noreply@anthropic.com>
ci: run the live e2e call test on pull requests
…in retry and TLS/SRTP config Adds barge-in primitives (stop_audio/send_audio block/non-block/speak sentence-chunking), a /transfer wrapper around baresip's menu module REFER command, per-call metadata (CallInfo/parse_sip_uri) with capped call history and DTMF capture, opt-in registration retry with backoff before handle_login_failure quits, and TLS CA file / SRTP module exposure in render_config and the BareSIP account line. All additions are additive with defaults matching prior behavior. Co-Authored-By: Claude Fable 5 <noreply@anthropic.com>
feat: interruptible playback, transfer, caller identity, login retry, TLS/SRTP
Adds baresipy.ivr (IVRNode/IVRSession/IVRPhone) for building phone-tree call flows with barge-in, retries, submenus, and transfer/hangup actions. Extends the HTTP gateway with /transfer, /stop_audio, /calls, and a per-call call_id carried in /ws/events payloads so events can be grouped by call. Co-Authored-By: Claude Fable 5 <noreply@anthropic.com>
feat: declarative IVR menus and gateway v2 (transfer, barge-in, call history)
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Human review requested!