# Asterisk Lab Config This folder contains the lab-grade configuration set for Track 9, Track 11, and Track 14: - `pjsip.conf` - `extensions.conf` - `manager.conf` - `http.conf` - `rtp.conf` These files are templates for a dedicated Linux VM running Asterisk 20 LTS. Before using them: 1. Replace default passwords and secrets. 2. Confirm the softphone source network. 3. Create the recording directory: - `/var/spool/asterisk/monitor/mvpcc` 4. Point `ASTERISK_QUEUE_MAP_JSON` in the platform to the matching queue code: - `voice_lab` - `voice_lab_ai` for the dedicated AI-first staging entrypoint - `voice_lab_ivr` for the dedicated IVR staging entrypoint 5. Configure operator extension mapping in bridge env for Track 11 call control: - `ASTERISK_OPERATOR_EXTENSION_MAP_JSON={"operator_a":"2001","operator_b":"2002"}` 6. Optional transfer map for queue-code based blind transfer: - `ASTERISK_TRANSFER_TARGET_MAP_JSON={"voice_lab":"2001","voice_lab_ai":"2001","voice_lab_ivr":"7200"}` 7. For IVR-over-Asterisk staging: - enable `ASTERISK_IVR_FASTAGI_ENABLED=1` - set `ASTERISK_IVR_FASTAGI_HOST=` - set `ASTERISK_IVR_FASTAGI_PORT=4573` - set `IVR_RUNTIME_TRUSTED_SERVICE_SUBJECTS=svc:asterisk-bridge` - provision the referenced `prompt_audio_key` files under the Asterisk sounds directory - if you want a higher-quality prompt from OpenAI TTS, generate it once and keep it on the server: - `python scripts/materialize_tts_prompt.py --text "Здравствуйте! Добро пожаловать в контакт-центр. Для русского языка нажмите 1. Қазақ тілі үшін 2 басыңыз." --output /usr/share/asterisk/sounds/en/mvpcc_ivr_language.wav --language ru` - `python scripts/materialize_tts_prompt.py --text "Для отдела продаж нажмите 1. Для службы поддержки нажмите 2." --output /usr/share/asterisk/sounds/en/mvpcc_ivr_menu_ru.wav --language ru` - `python scripts/materialize_tts_prompt.py --text "Сату бөлімі үшін 1 басыңыз. Қолдау қызметі үшін 2 басыңыз." --output /usr/share/asterisk/sounds/en/mvpcc_ivr_menu_kz.wav --language kk` - repeated runs reuse the server cache from `AI_VOICE_TTS_CACHE_DIR` instead of calling OpenAI again for the same text 8. For Track 14 browser softphone QA: - enable `ASTERISK_WEBRTC_ENABLED=1` - set `ASTERISK_WEBRTC_WS_URL=wss://:8089/ws` - configure `ASTERISK_BROWSER_SIP_MAP_JSON` - replace TLS placeholders in `http.conf` with a trusted cert/key pair 9. For Voice AI V1 lab wiring: - enable `AI_VOICE_ENABLED=1` - set `AI_VOICE_QUEUE_CONFIG_JSON={"voice_lab_ai":{"mode":"ai_first","agent_profile":"voice_support","handoff_queue_code":"voice_lab","language":"ru"}}` - set `AI_VOICE_RUNTIME_SERVICE_URL=http://ai-voice-runtime-service:8000` - keep `AI_VOICE_RUNTIME_TRUSTED_SERVICE_SUBJECTS=svc:ai-voice-runtime` - load `app_audiosocket.so` on the Asterisk VM before enabling the AI media path - set `MVPCC_AI_AUDIOSOCKET_SERVICE=:` in `extensions.conf` only after the AudioSocket listener is reachable Entrypoints in `extensions.conf`: - `7000`: human-only baseline. This path preserves the original Track 9/11 behavior. - `7100`: AI-first staging path. It emits the normal `MVPCCCallStarted` event with `QueueCode=voice_lab_ai`, then: - falls back to human dialing immediately if `MVPCC_AI_AUDIOSOCKET_SERVICE` is blank - otherwise starts `AudioSocket(,:)` for the media bridge - `7200`: IVR staging path. It emits the normal `MVPCCCallStarted` event with `QueueCode=voice_lab_ivr`, then: - hands control to FastAGI at `MVPCC_IVR_FASTAGI_HOSTPORT` - routes completed IVR sessions through `mvpcc-transfer` - falls back to the human baseline if FastAGI does not return a transfer target Important Voice AI note: - `AI_VOICE_RUNTIME_SERVICE_URL` is the HTTP control-plane endpoint for platform services. - `MVPCC_AI_AUDIOSOCKET_SERVICE` is a separate raw AudioSocket address for Asterisk media. - Do not point `MVPCC_AI_AUDIOSOCKET_SERVICE` at the runtime HTTP port. Default lab SIP accounts from `pjsip.conf`: - `1001` (caller) - `2001` (operator A) - `2002` (operator B) Track 14 note: - `2001` and `2002` are allowed to keep multiple contacts so browser registration can coexist with `MicroSIP/Zoiper` in QA.