Initial import with GitLab CI/CD and registry deploy flow

deployment/asterisk/Dockerfile

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+FROM ubuntu:24.04
+
+ENV DEBIAN_FRONTEND=noninteractive
+
+RUN apt-get update \
+    && apt-get install -y --no-install-recommends \
+        asterisk \
+        ca-certificates \
+        curl \
+        openssl \
+        python3 \
+        procps \
+        uuid-runtime \
+    && rm -rf /var/lib/apt/lists/*
+
+COPY deployment/asterisk/entrypoint.sh /usr/local/bin/mvpcc-asterisk-entrypoint.sh
+
+RUN chmod +x /usr/local/bin/mvpcc-asterisk-entrypoint.sh \
+    && mkdir -p /config-template /assets /etc/asterisk/keys /var/spool/asterisk/monitor/mvpcc
+
+ENTRYPOINT ["/usr/local/bin/mvpcc-asterisk-entrypoint.sh"]

deployment/asterisk/README.md

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+# 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=<bridge-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://<asterisk-host>: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=<media-host>:<port>` 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(<uuid>,<host>:<port>)` 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.

deployment/asterisk/entrypoint.sh

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+#!/usr/bin/env bash
+set -euo pipefail
+
+template_dir="/config-template"
+monitor_dir="/var/spool/asterisk/monitor/mvpcc"
+primary_sounds_dir="/usr/share/asterisk/sounds"
+legacy_sounds_dir="/var/lib/asterisk/sounds"
+keys_dir="/etc/asterisk/keys"
+
+password_1001="${ASTERISK_ENDPOINT_1001_PASSWORD:-${ASTERISK_AMI_SECRET:-ChangeMe-1001}}"
+password_2001="${ASTERISK_ENDPOINT_2001_PASSWORD:-${ASTERISK_AMI_SECRET:-ChangeMe-2001}}"
+password_2002="${ASTERISK_ENDPOINT_2002_PASSWORD:-${ASTERISK_AMI_SECRET:-ChangeMe-2002}}"
+ami_secret="${ASTERISK_AMI_SECRET:-ChangeMe-AmiSecret}"
+external_address="${ASTERISK_EXTERNAL_ADDRESS:-92.38.48.166}"
+
+mkdir -p /etc/asterisk "$monitor_dir" "$primary_sounds_dir" "$legacy_sounds_dir" "$keys_dir"
+
+for file in extensions.conf pjsip.conf manager.conf http.conf rtp.conf; do
+  cp "${template_dir}/${file}" "/etc/asterisk/${file}"
+done
+
+export ASTERISK_PASSWORD_1001="$password_1001"
+export ASTERISK_PASSWORD_2001="$password_2001"
+export ASTERISK_PASSWORD_2002="$password_2002"
+export ASTERISK_MANAGER_SECRET="$ami_secret"
+export ASTERISK_EXTERNAL_ADDRESS_VALUE="$external_address"
+
+python3 - <<'PY'
+from pathlib import Path
+import os
+
+replacements = {
+    "ChangeMe-1001": os.environ["ASTERISK_PASSWORD_1001"],
+    "ChangeMe-2001": os.environ["ASTERISK_PASSWORD_2001"],
+    "ChangeMe-2002": os.environ["ASTERISK_PASSWORD_2002"],
+    "ChangeMe-AmiSecret": os.environ["ASTERISK_MANAGER_SECRET"],
+    "92.38.48.166": os.environ["ASTERISK_EXTERNAL_ADDRESS_VALUE"],
+}
+
+for path in (
+    Path("/etc/asterisk/pjsip.conf"),
+    Path("/etc/asterisk/manager.conf"),
+):
+    text = path.read_text(encoding="utf-8")
+    for old, new in replacements.items():
+        text = text.replace(old, new)
+    path.write_text(text, encoding="utf-8")
+
+modules_path = Path("/etc/asterisk/modules.conf")
+if modules_path.exists():
+    text = modules_path.read_text(encoding="utf-8")
+    if "noload => chan_sip.so" not in text:
+        stanza = (
+            "; MVPCC uses PJSIP/WebRTC for browser softphones.\n"
+            "noload => chan_sip.so\n\n"
+        )
+        if "[global]" in text:
+            text = text.replace("[global]", f"{stanza}[global]", 1)
+        elif "[modules]" in text:
+            text = text.rstrip() + "\n\n" + stanza
+        else:
+            text = "[modules]\nautoload=yes\n\n" + stanza + text
+        modules_path.write_text(text, encoding="utf-8")
+PY
+
+if [ ! -f "${keys_dir}/asterisk.pem" ] || [ ! -f "${keys_dir}/asterisk.key" ]; then
+  openssl req \
+    -x509 \
+    -nodes \
+    -newkey rsa:2048 \
+    -keyout "${keys_dir}/asterisk.key" \
+    -out "${keys_dir}/asterisk.pem" \
+    -days 3650 \
+    -subj "/CN=${external_address}"
+fi
+
+if [ -f /assets/mvpcc_ivr_prompt.wav ]; then
+  for sounds_dir in "$primary_sounds_dir" "$legacy_sounds_dir"; do
+    cp /assets/mvpcc_ivr_prompt.wav "${sounds_dir}/mvpcc_ivr_prompt.wav"
+    mkdir -p "${sounds_dir}/en" "${sounds_dir}/ru" "${sounds_dir}/ru_RU"
+    cp /assets/mvpcc_ivr_prompt.wav "${sounds_dir}/en/mvpcc_ivr_prompt.wav"
+    cp /assets/mvpcc_ivr_prompt.wav "${sounds_dir}/ru/mvpcc_ivr_prompt.wav"
+    cp /assets/mvpcc_ivr_prompt.wav "${sounds_dir}/ru_RU/mvpcc_ivr_prompt.wav"
+  done
+fi
+
+if [ -d /assets/ivr ]; then
+  for sounds_dir in "$primary_sounds_dir" "$legacy_sounds_dir"; do
+    mkdir -p "${sounds_dir}/ivr"
+    cp -a /assets/ivr/. "${sounds_dir}/ivr/"
+  done
+fi
+
+chown -R asterisk:asterisk /etc/asterisk "$keys_dir" /var/spool/asterisk/monitor
+
+for sounds_dir in "$primary_sounds_dir" "$legacy_sounds_dir"; do
+  if [ -w "$sounds_dir" ]; then
+    chown -R asterisk:asterisk "$sounds_dir"
+  fi
+done
+
+chmod 0755 /var/spool/asterisk /var/spool/asterisk/monitor "$monitor_dir"
+
+exec /usr/sbin/asterisk -f -U asterisk -G asterisk -vvv

deployment/asterisk/extensions.conf

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+[globals]
+MVPCC_QUEUE_CODE=voice_lab
+MVPCC_AI_QUEUE_CODE=voice_lab_ai
+MVPCC_IVR_QUEUE_CODE=voice_lab_ivr
+MVPCC_IVR_FALLBACK_QUEUE_CODE=voice_lab
+MVPCC_MONITOR_DIR=/var/spool/asterisk/monitor/mvpcc
+MVPCC_HUMAN_DIAL_TARGETS=PJSIP/2001&PJSIP/2002
+MVPCC_AI_AUDIOSOCKET_SERVICE=127.0.0.1:9019
+MVPCC_IVR_FASTAGI_HOSTPORT=127.0.0.1:4573
+
+[from-softphones]
+exten => 7000,1,NoOp(MVPCC human baseline entrypoint)
+ same => n,Answer()
+ same => n,Set(MVPCC_QUEUE=${MVPCC_QUEUE_CODE})
+ same => n,Gosub(mvpcc-call-init,s,1(${EXTEN}))
+ same => n,Dial(${MVPCC_HUMAN_DIAL_TARGETS},40,U(mvpcc-connected^${MVPCC_CALL_ID}^${MVPCC_LINKED_ID}))
+ same => n,Hangup()
+
+exten => 7100,1,NoOp(MVPCC AI-first voice lab entrypoint)
+ same => n,Set(MVPCC_QUEUE=${IF($["${MVPCC_AI_QUEUE_OVERRIDE}" != ""]?${MVPCC_AI_QUEUE_OVERRIDE}:${MVPCC_AI_QUEUE_CODE})})
+ same => n,GotoIf($["${MVPCC_AI_REUSE_CALL}" = "1"]?ai-reuse,1)
+ same => n,Answer()
+ same => n,Gosub(mvpcc-call-init,s,1(${EXTEN}))
+ same => n,Goto(ai-audiosocket,1)
+
+exten => ai-reuse,1,NoOp(MVPCC AI handoff reusing active IVR call)
+ same => n,Goto(ai-audiosocket,1)
+
+exten => ai-audiosocket,1,NoOp(MVPCC AI audio bridge start)
+ same => n,GotoIf($["${MVPCC_AI_AUDIOSOCKET_SERVICE}" = ""]?ai-human-fallback,1)
+ same => n,Set(MVPCC_MEDIA_UUID=${SHELL(uuidgen | tr -d '\r\n')})
+ same => n,UserEvent(MVPCCAIAudioBridgeRequested,CallID:${MVPCC_CALL_ID},LinkedID:${MVPCC_LINKED_ID},QueueCode:${MVPCC_QUEUE},Extension:${MVPCC_ENTRY_EXTENSION},Context:${CONTEXT},MediaUUID:${MVPCC_MEDIA_UUID},UuidLen:${LEN(${MVPCC_MEDIA_UUID})},AudioSocketService:${MVPCC_AI_AUDIOSOCKET_SERVICE},Channel:${CHANNEL})
+ same => n,UserEvent(MVPCCAIAudioSocketAttempt,CallID:${MVPCC_CALL_ID},LinkedID:${MVPCC_LINKED_ID},QueueCode:${MVPCC_QUEUE},Extension:${MVPCC_ENTRY_EXTENSION},Context:${CONTEXT},MediaUUID:${MVPCC_MEDIA_UUID},UuidLen:${LEN(${MVPCC_MEDIA_UUID})},AudioSocketService:${MVPCC_AI_AUDIOSOCKET_SERVICE},Channel:${CHANNEL})
+ same => n,TryExec(AudioSocket(${MVPCC_MEDIA_UUID},${MVPCC_AI_AUDIOSOCKET_SERVICE}))
+ same => n,UserEvent(MVPCCAIAudioSocketResult,CallID:${MVPCC_CALL_ID},LinkedID:${MVPCC_LINKED_ID},QueueCode:${MVPCC_QUEUE},Extension:${MVPCC_ENTRY_EXTENSION},Context:${CONTEXT},MediaUUID:${MVPCC_MEDIA_UUID},UuidLen:${LEN(${MVPCC_MEDIA_UUID})},AudioSocketService:${MVPCC_AI_AUDIOSOCKET_SERVICE},TryExecStatus:${TRYSTATUS},Channel:${CHANNEL})
+ same => n,GotoIf($["${TRYSTATUS}" = "SUCCESS"]?ai-bridge-ended,1)
+ same => n,UserEvent(MVPCCAIFallbackToHuman,CallID:${MVPCC_CALL_ID},LinkedID:${MVPCC_LINKED_ID},QueueCode:${MVPCC_QUEUE},Extension:${MVPCC_ENTRY_EXTENSION},Context:${CONTEXT},Reason:audiosocket_failed,TryExecStatus:${TRYSTATUS},Channel:${CHANNEL})
+ same => n,Dial(${MVPCC_HUMAN_DIAL_TARGETS},40,U(mvpcc-connected^${MVPCC_CALL_ID}^${MVPCC_LINKED_ID}))
+ same => n,Hangup()
+
+exten => ai-bridge-ended,1,UserEvent(MVPCCAIAudioBridgeEnded,CallID:${MVPCC_CALL_ID},LinkedID:${MVPCC_LINKED_ID},QueueCode:${MVPCC_QUEUE},Extension:${MVPCC_ENTRY_EXTENSION},Context:${CONTEXT},MediaUUID:${MVPCC_MEDIA_UUID},AudioSocketService:${MVPCC_AI_AUDIOSOCKET_SERVICE},Channel:${CHANNEL})
+ same => n,Hangup()
+
+exten => ai-human-fallback,1,UserEvent(MVPCCAIFallbackToHuman,CallID:${MVPCC_CALL_ID},LinkedID:${MVPCC_LINKED_ID},QueueCode:${MVPCC_QUEUE},Extension:${MVPCC_ENTRY_EXTENSION},Context:${CONTEXT},Reason:audiosocket_service_missing,Channel:${CHANNEL})
+ same => n,Dial(${MVPCC_HUMAN_DIAL_TARGETS},40,U(mvpcc-connected^${MVPCC_CALL_ID}^${MVPCC_LINKED_ID}))
+ same => n,Hangup()
+
+exten => 7200,1,NoOp(MVPCC IVR staging entrypoint)
+ same => n,Answer()
+ same => n,Set(MVPCC_QUEUE=${MVPCC_IVR_QUEUE_CODE})
+ same => n,Set(MVPCC_IVR_FALLBACK_QUEUE_CODE=${GLOBAL(MVPCC_IVR_FALLBACK_QUEUE_CODE)})
+ same => n,Gosub(mvpcc-call-init,s,1(${EXTEN}))
+ same => n,Set(MVPCC_IVR_RESULT=started)
+ same => n,Set(MVPCC_IVR_TARGET_QUEUE_CODE=)
+ same => n,Set(MVPCC_IVR_TARGET_EXTENSION=)
+ same => n,Set(MVPCC_IVR_SESSION_ID=)
+ same => n,GotoIf($["${MVPCC_IVR_FASTAGI_HOSTPORT}" = ""]?ivr-human-fallback,1)
+ same => n,AGI(agi://${MVPCC_IVR_FASTAGI_HOSTPORT}/ivr,${MVPCC_CALL_ID},${MVPCC_LINKED_ID},${MVPCC_QUEUE},${MVPCC_IVR_FALLBACK_QUEUE_CODE},${EXTEN})
+ same => n,GotoIf($["${MVPCC_IVR_TARGET_EXTENSION}" = ""]?ivr-human-fallback,1)
+ same => n,Goto(mvpcc-transfer,${MVPCC_IVR_TARGET_EXTENSION},1)
+
+exten => ivr-human-fallback,1,UserEvent(MVPCCIVRFallbackToHuman,CallID:${MVPCC_CALL_ID},LinkedID:${MVPCC_LINKED_ID},QueueCode:${MVPCC_QUEUE},FallbackQueueCode:${MVPCC_IVR_FALLBACK_QUEUE_CODE},Extension:${MVPCC_ENTRY_EXTENSION},Result:${MVPCC_IVR_RESULT},TargetQueueCode:${MVPCC_IVR_TARGET_QUEUE_CODE},TargetExtension:${MVPCC_IVR_TARGET_EXTENSION},Channel:${CHANNEL})
+ same => n,Dial(${MVPCC_HUMAN_DIAL_TARGETS},40,U(mvpcc-connected^${MVPCC_CALL_ID}^${MVPCC_LINKED_ID}))
+ same => n,Hangup()
+
+exten => h,1,StopMixMonitor()
+ same => n,UserEvent(MVPCCCallEnded,CallID:${MVPCC_CALL_ID},LinkedID:${MVPCC_LINKED_ID},HangupCause:${HANGUPCAUSE},DurationSeconds:${CDR(duration)})
+ same => n,UserEvent(MVPCCRecordingReady,CallID:${MVPCC_CALL_ID},LinkedID:${MVPCC_LINKED_ID},RemotePath:${MONITOR_FILENAME},FileName:${MVPCC_RECORD_FILE},MimeType:audio/wav,DurationSeconds:${CDR(duration)})
+
+[mvpcc-call-init]
+exten => s,1,Set(MVPCC_ENTRY_EXTENSION=${ARG1})
+ same => n,Set(MVPCC_CALL_ID=${UNIQUEID})
+ same => n,Set(MVPCC_LINKED_ID=${CHANNEL(linkedid)})
+ same => n,Set(MVPCC_RECORD_FILE=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${UNIQUEID}.wav)
+ same => n,Set(MONITOR_FILENAME=${MVPCC_MONITOR_DIR}/${MVPCC_RECORD_FILE})
+ same => n,UserEvent(MVPCCCallStarted,CallID:${MVPCC_CALL_ID},LinkedID:${MVPCC_LINKED_ID},CallerNumber:${CALLERID(num)},CallerName:${CALLERID(name)},QueueCode:${MVPCC_QUEUE},Extension:${MVPCC_ENTRY_EXTENSION},Context:${CONTEXT},Direction:inbound,Channel:${CHANNEL})
+ same => n,MixMonitor(${MONITOR_FILENAME},a)
+ same => n,Return()
+
+[mvpcc-connected]
+exten => s,1,Set(MVPCC_OPERATOR=${IF($["${ARG3}" != ""]?${ARG3}:${CHANNEL(peername)})})
+ same => n,Set(MVPCC_OPERATOR=${IF($["${MVPCC_OPERATOR}" = ""]?${EXTEN}:${MVPCC_OPERATOR})})
+ same => n,UserEvent(MVPCCOperatorConnected,CallID:${ARG1},LinkedID:${ARG2},OperatorExtension:${MVPCC_OPERATOR},Context:${CONTEXT},Channel:${CHANNEL},ClaimedByUser:${MVPCC_OPERATOR})
+ same => n,Return()
+
+[mvpcc-claim]
+exten => _X.,1,NoOp(Claimed call routed to operator ${EXTEN})
+ same => n,Set(MVPCC_CALL_ID=${CHANNEL(linkedid)})
+ same => n,Set(MVPCC_LINKED_ID=${CHANNEL(linkedid)})
+ same => n,Dial(PJSIP/${EXTEN},40,U(mvpcc-connected^${MVPCC_CALL_ID}^${MVPCC_LINKED_ID}^${EXTEN}))
+ same => n,Hangup()
+
+[mvpcc-transfer]
+exten => 7000,1,NoOp(Queue transfer target ${EXTEN})
+ same => n,Dial(${MVPCC_HUMAN_DIAL_TARGETS},40)
+ same => n,Hangup()
+
+exten => 7100,1,NoOp(Queue transfer target ${EXTEN})
+ same => n,Goto(from-softphones,7100,1)
+
+exten => 7200,1,NoOp(Queue transfer target ${EXTEN})
+ same => n,Goto(from-softphones,7200,1)
+
+exten => _X.,1,NoOp(Blind transfer target ${EXTEN})
+ same => n,Dial(PJSIP/${EXTEN},40,U(mvpcc-connected^${MVPCC_CALL_ID}^${MVPCC_LINKED_ID}^${EXTEN}))
+ same => n,GotoIf($["${DIALSTATUS}" = "ANSWER"]?transfer-finished,1)
+ same => n,GotoIf($["${MVPCC_IVR_FALLBACK_QUEUE_CODE}" = ""]?transfer-finished,1)
+ same => n,Set(MVPCC_IVR_RESULT=fallback_transfer_unavailable)
+ same => n,Goto(from-softphones,ivr-human-fallback,1)
+
+exten => transfer-finished,1,Hangup()
+
+exten => h,1,StopMixMonitor()
+ same => n,UserEvent(MVPCCCallEnded,CallID:${MVPCC_CALL_ID},LinkedID:${MVPCC_LINKED_ID},HangupCause:${HANGUPCAUSE},DurationSeconds:${CDR(duration)})
+ same => n,UserEvent(MVPCCRecordingReady,CallID:${MVPCC_CALL_ID},LinkedID:${MVPCC_LINKED_ID},RemotePath:${MONITOR_FILENAME},FileName:${MVPCC_RECORD_FILE},MimeType:audio/wav,DurationSeconds:${CDR(duration)})
+
+[mvpcc-operators]
+exten => _X.,1,Dial(PJSIP/${EXTEN},30)
+ same => n,Hangup()

deployment/asterisk/http.conf

@@ -0,0 +1,12 @@
+[general]
+enabled=yes
+enablestatic=yes
+bindaddr=0.0.0.0
+bindport=8088
+
+; QA-only TLS placeholders for direct browser SIP over WebSocket.
+; Replace with your actual trusted certificate and key paths before enabling WSS.
+tlsenable=yes
+tlsbindaddr=0.0.0.0:8089
+tlscertfile=/etc/asterisk/keys/asterisk.pem
+tlsprivatekey=/etc/asterisk/keys/asterisk.key

deployment/asterisk/manager.conf

@@ -0,0 +1,11 @@
+[general]
+enabled = yes
+port = 5038
+bindaddr = 0.0.0.0
+
+[mvpcc]
+secret = ChangeMe-AmiSecret
+deny = 0.0.0.0/0.0.0.0
+permit = 0.0.0.0/0.0.0.0
+read = system,call,log,verbose,command,agent,user,dtmf,reporting,cdr,dialplan
+write = command,call,originate,agent,user

deployment/asterisk/pjsip.conf

@@ -0,0 +1,84 @@
+[transport-udp]
+type=transport
+protocol=udp
+bind=0.0.0.0
+external_signaling_address=92.38.48.166
+external_media_address=92.38.48.166
+
+[transport-wss]
+type=transport
+protocol=wss
+bind=0.0.0.0
+external_signaling_address=92.38.48.166
+external_media_address=92.38.48.166
+
+[softphone-endpoint](!)
+type=endpoint
+disallow=all
+allow=opus,ulaw,alaw
+dtmf_mode=auto_info
+direct_media=no
+rtp_symmetric=yes
+force_rport=yes
+rewrite_contact=yes
+webrtc=yes
+ice_support=yes
+rtcp_mux=yes
+use_avpf=yes
+media_encryption=dtls
+dtls_auto_generate_cert=yes
+dtls_setup=actpass
+media_use_received_transport=yes
+media_address=92.38.48.166
+
+[caller-endpoint](!,softphone-endpoint)
+context=from-softphones
+
+[operator-endpoint](!,softphone-endpoint)
+context=mvpcc-operators
+
+[1001](caller-endpoint)
+auth=1001-auth
+aors=1001
+
+[1001-auth]
+type=auth
+auth_type=userpass
+username=1001
+password=ChangeMe-1001
+
+[1001]
+type=aor
+max_contacts=2
+
+[2001](operator-endpoint)
+auth=2001-auth
+aors=2001
+
+[2001-auth]
+type=auth
+auth_type=userpass
+username=2001
+password=ChangeMe-2001
+
+[2001]
+type=aor
+max_contacts=1
+remove_existing=yes
+support_path=yes
+
+[2002](operator-endpoint)
+auth=2002-auth
+aors=2002
+
+[2002-auth]
+type=auth
+auth_type=userpass
+username=2002
+password=ChangeMe-2002
+
+[2002]
+type=aor
+max_contacts=1
+remove_existing=yes
+support_path=yes

deployment/asterisk/rtp.conf

@@ -0,0 +1,5 @@
+[general]
+rtpstart=10000
+rtpend=20000
+icesupport=true
+stunaddr=stun.l.google.com:19302