Senad Uka 714aa1782b M4: voice pipeline (software-only telephony) ...

- AudioSocket server (asyncio TCP, Asterisk wire protocol) bridging calls
  into the same M2 agent used by chat
- Call session engine: greeting → energy-VAD utterance collection → STT →
  agent turn → TTS playback with barge-in (120ms of caller speech stops
  playback); inactivity + max-duration guards; unintelligible audio reaches
  the agent as '[nerazumljivo]' so the two-attempt rule stays in the prompt
- STTProvider (faster-whisper, lang hint 'sr', GPU/CPU via env) and
  TTSProvider (Azure Neural raw-8k PCM / ElevenLabs Flash) + test fakes
- Recording (mixed caller+agent WAV), transcripts, per-turn latency trace,
  metering via record_agent_call (§12)
- Internal dialplan API: /internal/calls/register (ring targets computed from
  working hours + ring settings §5.2), /answered (human outcome, no minutes),
  /hangup (abandoned → missed-call SMS §5.5); shared-token auth
- Asterisk config generator (pjsip.conf + extensions.conf from DB): worker
  endpoints, per-tenant test-caller endpoint, register→Dial→AudioSocket
  dialplan with h-extension reporting — validated against a real Asterisk 20
  container (modules, dialplan, endpoints all load)
- docker-compose 'voice' profile (voice server + Asterisk), Dockerfile.voice,
  docs/VOICE_TESTING.md softphone runbook
- 16 new tests incl. full fake-call e2e and a real-TCP AudioSocket wire test

Co-Authored-By: Claude Fable 5 <noreply@anthropic.com>

2026-07-11 11:05:56 +02:00

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