first commit

.env.example

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+# ==========================================================
+# فایل تنظیمات مرکزی پروژه چت‌بات تلفنی
+# این تنها فایلی است که باید برای راه‌اندازی ادیت کنی.
+# بعد از تغییر، کافیه: docker compose up -d --build
+# ==========================================================
+
+# ---------- Ollama ----------
+OLLAMA_URL=http://OLLAMA_HOST:11434/api/generate
+OLLAMA_MODEL=llama3
+
+# ---------- STT (faster-whisper) ----------
+WHISPER_MODEL=medium
+WHISPER_DEVICE=cpu
+
+# ---------- TTS (Piper) ----------
+PIPER_VOICE=/models/piper/fa_IR-model.onnx
+
+# ---------- ترانک مرکز تماس شماره ۱ (SIP Trunk) ----------
+TRUNK1_NAME=callcenter1
+TRUNK1_HOST=CALLCENTER1_IP
+TRUNK1_USERNAME=USERNAME
+TRUNK1_PASSWORD=CALLCENTER1_PASS
+
+# ---------- ترانک مرکز تماس شماره ۲ (اختیاری - اگر نداری خالی بذار) ----------
+TRUNK2_NAME=
+TRUNK2_HOST=
+TRUNK2_USERNAME=
+TRUNK2_PASSWORD=
+
+# ---------- SIP User داخلی (برای تست با سافت‌فون) ----------
+TEST_USER_EXTEN=1000
+TEST_USER_PASSWORD=CHANGE_ME_STRONG_PASSWORD

.gitignore

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+.env
+models/
+asterisk/tmp/

agi-scripts/voicebot.py

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+#!/usr/bin/env python3
+# -*- coding: utf-8 -*-
+
+import os
+import uuid
+import requests
+from asterisk.agi import AGI
+
+
+TMP_DIR = "/tmp/voicebot"
+
+STT_URL = "http://127.0.0.1:5001/transcribe"
+TTS_URL = "http://127.0.0.1:5002/synthesize"
+
+OLLAMA_URL = "http://127.0.0.1:11434/api/generate"
+OLLAMA_MODEL = "qwen3:14b"
+
+
+
+def log(agi, msg):
+    agi.verbose(
+        "VOICEBOT: " + str(msg),
+        1
+    )
+
+
+
+def tts_generate(agi, text, name):
+
+    try:
+
+        log(
+            agi,
+            "TTS request: " + text
+        )
+
+
+        r = requests.post(
+            TTS_URL,
+            json={
+                "text": text
+            },
+            timeout=60
+        )
+
+        r.raise_for_status()
+
+
+        path = TMP_DIR + "/" + name + ".wav"
+
+
+        with open(path, "wb") as f:
+            f.write(r.content)
+
+
+        log(
+            agi,
+            "TTS created " + path
+        )
+
+
+        return path
+
+
+    except Exception as e:
+
+        log(
+            agi,
+            "TTS ERROR " + str(e)
+        )
+
+        return None
+
+
+
+def play_file(agi, path):
+
+    if not path:
+        return
+
+
+    try:
+
+        file = path.replace(
+            ".wav",
+            ""
+        )
+
+
+        agi.appexec(
+            "Playback",
+            file
+        )
+
+
+    except Exception as e:
+
+        log(
+            agi,
+            "PLAY ERROR " + str(e)
+        )
+
+
+
+
+def record_audio(agi, callid):
+
+
+    filename = TMP_DIR + "/" + callid + "_record"
+
+
+    log(
+        agi,
+        "Recording user voice"
+    )
+
+
+    try:
+
+
+        result = agi.send_command(
+
+            f'RECORD FILE "{filename}" wav "#" 10000 0 s=5'
+
+        )
+
+
+        log(
+            agi,
+            "Record result: " + str(result)
+        )
+
+
+    except Exception as e:
+
+
+        log(
+            agi,
+            "Record ERROR " + str(e)
+        )
+
+        return None
+
+
+
+    wav = filename + ".wav"
+
+
+    if os.path.exists(wav):
+
+
+        size=os.path.getsize(wav)
+
+
+        log(
+            agi,
+            f"Recording saved {wav} size={size}"
+        )
+
+
+        return wav
+
+
+
+    log(
+        agi,
+        "Recording file missing"
+    )
+
+
+    return None
+
+
+
+
+
+def stt_process(agi,wav):
+
+
+    try:
+
+
+        log(
+            agi,
+            "Sending audio to STT"
+        )
+
+
+        with open(wav,"rb") as f:
+
+
+            r=requests.post(
+
+                STT_URL,
+
+                files={
+                    "audio":f
+                },
+
+                timeout=120
+
+            )
+
+
+        data=r.json()
+
+
+        text=data.get(
+            "text",
+            ""
+        )
+
+
+        log(
+            agi,
+            "STT text: " + text
+        )
+
+
+        return text
+
+
+
+    except Exception as e:
+
+
+        log(
+            agi,
+            "STT ERROR " + str(e)
+        )
+
+
+        return ""
+
+
+
+
+
+def ask_ai(agi,text):
+
+
+    try:
+
+
+        log(
+            agi,
+            "AI request: " + text
+        )
+
+
+        prompt = """
+تو دستیار تلفنی هستی.
+کوتاه و طبیعی جواب بده.
+حداکثر دو جمله.
+
+کاربر:
+""" + text
+
+
+
+        r=requests.post(
+
+            OLLAMA_URL,
+
+            json={
+
+                "model":OLLAMA_MODEL,
+
+                "prompt":prompt,
+
+                "stream":False,
+
+                "options":{
+                    "num_predict":80,
+                    "temperature":0.3
+                }
+
+            },
+
+            timeout=180
+
+        )
+
+
+        data=r.json()
+
+
+        answer=data.get(
+            "response",
+            ""
+        )
+
+
+        log(
+            agi,
+            "AI response: " + answer
+        )
+
+
+        return answer
+
+
+
+    except Exception as e:
+
+
+        log(
+            agi,
+            "AI ERROR " + str(e)
+        )
+
+
+        return "ارتباط با هوش مصنوعی برقرار نشد."
+
+
+
+
+
+def main():
+
+
+    os.makedirs(
+        TMP_DIR,
+        exist_ok=True
+    )
+
+
+    agi=AGI()
+
+
+    callid=str(uuid.uuid4())[:8]
+
+
+    log(
+        agi,
+        "Call started " + callid
+    )
+
+
+    try:
+
+
+        agi.answer()
+
+
+
+        welcome=tts_generate(
+
+            agi,
+
+            "سلام، من دستیار صوتی هوشمند هستم. بفرمایید.",
+
+            callid+"_welcome"
+
+        )
+
+
+        play_file(
+            agi,
+            welcome
+        )
+
+
+
+        wav=record_audio(
+            agi,
+            callid
+        )
+
+
+        if not wav:
+
+            log(
+                agi,
+                "No recording"
+            )
+
+            return
+
+
+
+        text=stt_process(
+            agi,
+            wav
+        )
+
+
+
+        if text:
+
+            answer=ask_ai(
+                agi,
+                text
+            )
+
+        else:
+
+            answer="صدای شما را دریافت نکردم."
+
+
+
+        reply=tts_generate(
+
+            agi,
+
+            answer,
+
+            callid+"_reply"
+
+        )
+
+
+        play_file(
+            agi,
+            reply
+        )
+
+
+        log(
+            agi,
+            "Call completed"
+        )
+
+
+
+    except Exception as e:
+
+
+        log(
+            agi,
+            "MAIN ERROR " + str(e)
+        )
+
+
+    finally:
+
+        agi.hangup()
+
+
+
+
+
+if __name__ == "__main__":
+
+    main()

asterisk/Dockerfile

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+FROM andrius/asterisk:latest
+
+RUN apt-get update && apt-get install -y \
+    python3 \
+    python3-pip \
+    gettext-base \
+    curl \
+    file \
+    sox \
+    ffmpeg \
+    && rm -rf /var/lib/apt/lists/*
+
+RUN pip3 install --break-system-packages \
+    requests \
+    pyst2
+
+RUN mkdir -p \
+    /tmp/voicebot \
+    /var/lib/asterisk/sounds/custom \
+    /etc/asterisk-templates
+
+COPY templates/ /etc/asterisk-templates/
+COPY entrypoint.sh /entrypoint.sh
+
+RUN chmod +x /entrypoint.sh
+
+EXPOSE 5060/udp
+EXPOSE 5060/tcp
+EXPOSE 10000-20000/udp
+
+ENTRYPOINT ["/entrypoint.sh"]

asterisk/config/agi.conf

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+[general]
+enabled = yes
+agidir => /var/lib/asterisk/agi-bin

asterisk/config/extensions.conf

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+[general]
+static=yes
+writeprotect=no
+
+
+; ==========================================================
+; Incoming calls from SIP Trunk
+; ==========================================================
+
+[from-trunk]
+
+; Incoming call without DID
+exten => s,1,NoOp(Incoming call from trunk - CallerID: ${CALLERID(num)})
+ same => n,Answer()
+ same => n,Wait(1)
+ same => n,AGI(voicebot.py)
+ same => n,Hangup()
+
+
+; Incoming call with DID
+exten => _X.,1,NoOp(Incoming DID call: ${EXTEN} - CallerID: ${CALLERID(num)})
+ same => n,Answer()
+ same => n,Wait(1)
+ same => n,AGI(voicebot.py)
+ same => n,Hangup()
+
+
+
+; ==========================================================
+; Internal test extension
+; Used with Zoiper / Linphone for testing
+; ==========================================================
+
+[from-internal]
+
+; Match any dialed extension
+exten => _X.,1,NoOp(Test call from internal extension ${EXTEN})
+ same => n,Answer()
+ same => n,Wait(1)
+ same => n,AGI(voicebot.py)
+ same => n,Hangup()

asterisk/config/modules.conf

@@ -0,0 +1,5 @@
+[modules]
+autoload=yes
+
+; در صورتی که نمی‌خوای chan_sip قدیمی لود بشه (توصیه می‌شه فقط pjsip استفاده کنی)
+noload => chan_sip.so

asterisk/config/pjsip.conf

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+[global]
+type=global
+user_agent=Pline-AI
+
+[transport-udp]
+type=transport
+protocol=udp
+bind=0.0.0.0:5060
+
+; ======================
+; Test Extension
+; ======================
+
+[1000]
+type=endpoint
+context=from-internal
+disallow=all
+allow=ulaw,alaw
+auth=1000-auth
+aors=1000
+
+[1000-auth]
+type=auth
+auth_type=userpass
+username=1000
+password=1000
+
+[1000]
+type=aor
+max_contacts=5
+[121-reg]
+type=registration
+transport=transport-udp
+outbound_auth=121-auth
+server_uri=sip:192.168.102.253
+client_uri=sip:121@192.168.102.253
+retry_interval=60
+expiration=3600
+
+[121-auth]
+type=auth
+auth_type=userpass
+username=121
+password=121
+
+[121]
+type=aor
+contact=sip:192.168.102.253
+qualify_frequency=60
+
+[121]
+type=endpoint
+context=from-trunk
+transport=transport-udp
+disallow=all
+allow=ulaw,alaw
+outbound_auth=121-auth
+aors=121
+
+; --- اضافه کردن این بخش ---
+[121-identify]
+type=identify
+endpoint=121
+match=192.168.102.253

asterisk/entrypoint.sh

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+#!/bin/sh
+set -e
+
+TEMPLATES_DIR="/etc/asterisk-templates"
+CONFIG_DIR="/etc/asterisk"
+
+echo "[entrypoint] generating asterisk configuration..."
+
+# کپی فایل‌های ثابت
+cp "$TEMPLATES_DIR/agi.conf" "$CONFIG_DIR/agi.conf"
+cp "$TEMPLATES_DIR/modules.conf" "$CONFIG_DIR/modules.conf"
+
+# تولید extensions.conf
+envsubst '${TEST_USER_EXTEN}' \
+    < "$TEMPLATES_DIR/extensions.conf.template" \
+    > "$CONFIG_DIR/extensions.conf"
+
+# تولید pjsip.conf پایه (بدون ترانک)
+envsubst '${TEST_USER_EXTEN} ${TEST_USER_PASSWORD}' \
+    < "$TEMPLATES_DIR/pjsip-base.conf.template" \
+    > "$CONFIG_DIR/pjsip.conf"
+
+# تابع افزودن ترانک
+add_trunk() {
+    TRUNK_NAME=$1 TRUNK_HOST=$2 TRUNK_USERNAME=$3 TRUNK_PASSWORD=$4 \
+    envsubst '${TRUNK_NAME} ${TRUNK_HOST} ${TRUNK_USERNAME} ${TRUNK_PASSWORD}' \
+    < "$TEMPLATES_DIR/pjsip-trunk.conf.template" \
+    >> "$CONFIG_DIR/pjsip.conf"
+}
+
+# اضافه کردن ترانک ۱
+add_trunk "$TRUNK1_NAME" "$TRUNK1_HOST" "$TRUNK1_USERNAME" "$TRUNK1_PASSWORD"
+
+# اضافه کردن ترانک ۲ (اگر وجود داشت)
+if [ -n "$TRUNK2_HOST" ]; then
+    echo "[entrypoint] adding second trunk..."
+    add_trunk "$TRUNK2_NAME" "$TRUNK2_HOST" "$TRUNK2_USERNAME" "$TRUNK2_PASSWORD"
+fi
+
+echo "[entrypoint] configuration generated"
+exec asterisk -f -vvv

asterisk/templates/agi.conf

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+[general]
+enabled = yes
+agidir => /var/lib/asterisk/agi-bin

asterisk/templates/extensions.conf.template

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+[general]
+static=yes
+writeprotect=no
+
+
+; ==========================================================
+; Incoming calls from SIP Trunk
+; ==========================================================
+
+[from-trunk]
+
+; Incoming call without DID
+exten => s,1,NoOp(Incoming call from trunk - CallerID: ${CALLERID(num)})
+ same => n,Answer()
+ same => n,Wait(1)
+ same => n,AGI(voicebot.py)
+ same => n,Hangup()
+
+
+; Incoming call with DID
+exten => _X.,1,NoOp(Incoming DID call: ${EXTEN} - CallerID: ${CALLERID(num)})
+ same => n,Answer()
+ same => n,Wait(1)
+ same => n,AGI(voicebot.py)
+ same => n,Hangup()
+
+
+
+; ==========================================================
+; Internal test extension
+; Used with Zoiper / Linphone for testing
+; ==========================================================
+
+[from-internal]
+
+; Match any dialed extension
+exten => _X.,1,NoOp(Test call from internal extension ${EXTEN})
+ same => n,Answer()
+ same => n,Wait(1)
+ same => n,AGI(voicebot.py)
+ same => n,Hangup()

asterisk/templates/modules.conf

@@ -0,0 +1,5 @@
+[modules]
+autoload=yes
+
+; در صورتی که نمی‌خوای chan_sip قدیمی لود بشه (توصیه می‌شه فقط pjsip استفاده کنی)
+noload => chan_sip.so

asterisk/templates/pjsip-base.conf.template

@@ -0,0 +1,30 @@
+[global]
+type=global
+user_agent=Pline-AI
+
+[transport-udp]
+type=transport
+protocol=udp
+bind=0.0.0.0:5060
+
+; ======================
+; Test Extension
+; ======================
+
+[${TEST_USER_EXTEN}]
+type=endpoint
+context=from-internal
+disallow=all
+allow=ulaw,alaw
+auth=${TEST_USER_EXTEN}-auth
+aors=${TEST_USER_EXTEN}
+
+[${TEST_USER_EXTEN}-auth]
+type=auth
+auth_type=userpass
+username=${TEST_USER_EXTEN}
+password=${TEST_USER_PASSWORD}
+
+[${TEST_USER_EXTEN}]
+type=aor
+max_contacts=5

asterisk/templates/pjsip-trunk.conf.template

@@ -0,0 +1,34 @@
+[${TRUNK_NAME}-reg]
+type=registration
+transport=transport-udp
+outbound_auth=${TRUNK_NAME}-auth
+server_uri=sip:${TRUNK_HOST}
+client_uri=sip:${TRUNK_USERNAME}@${TRUNK_HOST}
+retry_interval=60
+expiration=3600
+
+[${TRUNK_NAME}-auth]
+type=auth
+auth_type=userpass
+username=${TRUNK_USERNAME}
+password=${TRUNK_PASSWORD}
+
+[${TRUNK_NAME}]
+type=aor
+contact=sip:${TRUNK_HOST}
+qualify_frequency=60
+
+[${TRUNK_NAME}]
+type=endpoint
+context=from-trunk
+transport=transport-udp
+disallow=all
+allow=ulaw,alaw
+outbound_auth=${TRUNK_NAME}-auth
+aors=${TRUNK_NAME}
+
+; --- اضافه کردن این بخش ---
+[${TRUNK_NAME}-identify]
+type=identify
+endpoint=${TRUNK_NAME}
+match=${TRUNK_HOST}

docker-compose.yml

@@ -0,0 +1,65 @@
+services:
+  asterisk:
+    image: pline-ai/asterisk:latest
+    build:
+      context: ./asterisk
+    container_name: pline-ai-asterisk
+    network_mode: host
+    env_file:
+      - .env
+    environment:
+      STT_URL: http://127.0.0.1:5001/transcribe
+      TTS_URL: http://127.0.0.1:5002/synthesize
+    volumes:
+      - ./asterisk/config:/etc/asterisk
+      - ./asterisk/templates:/etc/asterisk-templates
+      - ./agi-scripts:/var/lib/asterisk/agi-bin
+      - ./asterisk/sounds:/var/lib/asterisk/sounds/custom
+      - voice_storage:/tmp/voicebot  # استفاده از volume مشترک
+    restart: unless-stopped
+    depends_on:
+      - stt-service
+      - tts-service
+
+  stt-service:
+    image: pline-ai/stt:latest
+    build:
+      context: ./stt
+    container_name: pline-ai-stt
+    network_mode: host
+    env_file:
+      - .env
+    environment:
+      PORT: 5001
+      WHISPER_MODEL: /models/whisper
+      WHISPER_DEVICE: cuda
+      WHISPER_COMPUTE_TYPE: float16
+    volumes:
+      - ./models/whisper:/models/whisper
+    deploy:
+      resources:
+        reservations:
+          devices:
+            - driver: nvidia
+              count: 1
+              capabilities:
+                - gpu
+    restart: unless-stopped
+
+  tts-service:
+    image: pline-ai/tts:latest
+    build:
+      context: ./tts
+    container_name: pline-ai-tts
+    network_mode: host
+    env_file:
+      - .env
+    environment:
+      PORT: 5002
+    volumes:
+      - ./models/piper:/models/piper
+      - voice_storage:/tmp/voicebot  # اضافه شده برای دسترسی به مسیر مشترک
+    restart: unless-stopped
+
+volumes:
+  voice_storage:  # تعریف حجم مشترک

stt/Dockerfile

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+FROM nvidia/cuda:12.4.1-runtime-ubuntu22.04
+
+ENV DEBIAN_FRONTEND=noninteractive
+
+RUN apt-get update && apt-get install -y \
+    python3 \
+    python3-pip \
+    ffmpeg \
+    && rm -rf /var/lib/apt/lists/*
+
+WORKDIR /app
+
+COPY requirements.txt .
+
+RUN pip3 install --no-cache-dir -r requirements.txt
+
+COPY app.py .
+
+EXPOSE 5001
+
+CMD ["python3", "app.py"]

stt/app.py

@@ -0,0 +1,78 @@
+# -*- coding: utf-8 -*-
+import os
+import tempfile
+from flask import Flask, request, jsonify
+from faster_whisper import WhisperModel
+
+# ==========================================================
+# Configuration
+# ==========================================================
+MODEL_PATH = os.environ.get("WHISPER_MODEL", "/models/whisper")
+DEVICE = os.environ.get("WHISPER_DEVICE", "cuda")
+COMPUTE_TYPE = "float16" if DEVICE == "cuda" else "int8"
+PORT = int(os.environ.get("PORT", 5001))
+
+app = Flask(__name__)
+
+# ==========================================================
+# Load Whisper Model
+# ==========================================================
+print(f"[stt] Loading model from {MODEL_PATH} on {DEVICE} ({COMPUTE_TYPE})")
+try:
+    model = WhisperModel(MODEL_PATH, device=DEVICE, compute_type=COMPUTE_TYPE)
+    print("[stt] Whisper model loaded successfully")
+except Exception as e:
+    print(f"[stt] Fatal error loading model: {e}")
+    exit(1)
+
+# ==========================================================
+# Speech To Text Endpoint
+# ==========================================================
+@app.route("/transcribe", methods=["POST"])
+def transcribe():
+    if "audio" not in request.files:
+        return jsonify({"error": "audio file missing"}), 400
+
+    audio_file = request.files["audio"]
+    
+    # استفاده از NamedTemporaryFile با مدیریت بهتر برای محیط داکر
+    tmp_file = tempfile.NamedTemporaryFile(delete=False, suffix=".wav")
+    audio_path = tmp_file.name
+    tmp_file.close()
+
+    try:
+        audio_file.save(audio_path)
+
+        # اجرای اینفرنس با تنظیمات بهینه شده
+        segments, info = model.transcribe(
+            audio_path,
+            language="fa",
+            beam_size=10,
+            vad_filter=True,
+            vad_parameters=dict(min_silence_duration_ms=500) # بهبود تشخیص سکوت
+        )
+
+        text = " ".join([segment.text.strip() for segment in segments])
+        
+        return jsonify({
+            "text": text.strip(),
+            "language": info.language
+        })
+
+    except Exception as e:
+        print(f"[stt] Error during transcription: {e}")
+        return jsonify({"error": str(e)}), 500
+
+    finally:
+        if os.path.exists(audio_path):
+            os.remove(audio_path)
+
+# ==========================================================
+# Health Check
+# ==========================================================
+@app.route("/health", methods=["GET"])
+def health():
+    return jsonify({"status": "ok"})
+
+if __name__ == "__main__":
+    app.run(host="0.0.0.0", port=PORT)

stt/requirements.txt

@@ -0,0 +1,5 @@
+flask==3.0.3
+faster-whisper==1.0.3
+ctranslate2
+requests
+numpy

tts/Dockerfile

@@ -0,0 +1,14 @@
+FROM python:3.11-slim
+
+RUN apt-get update && apt-get install -y \
+    wget ffmpeg \
+    && rm -rf /var/lib/apt/lists/*
+
+WORKDIR /app
+COPY requirements.txt .
+RUN pip install --no-cache-dir -r requirements.txt
+
+COPY app.py .
+
+EXPOSE 5002
+CMD ["python3", "app.py"]

tts/app.py

@@ -0,0 +1,18 @@
+import os, subprocess, tempfile
+from flask import Flask, request, send_file
+
+app = Flask(__name__)
+PIPER_VOICE = os.environ.get("PIPER_VOICE", "/models/piper/fa_IR-model.onnx")
+
+@app.route("/synthesize", methods=["POST"])
+def synthesize():
+    text = request.get_json().get("text", "")
+    p_out = tempfile.NamedTemporaryFile(suffix=".wav", delete=False).name
+    a_out = tempfile.NamedTemporaryFile(suffix=".wav", delete=False).name
+    
+    subprocess.run(["piper", "--model", PIPER_VOICE, "--output_file", p_out], input=text.encode("utf-8"), capture_output=True)
+    subprocess.run(["ffmpeg", "-y", "-i", p_out, "-ar", "8000", "-ac", "1", "-acodec", "pcm_s16le", "-f", "wav", a_out], stdout=subprocess.DEVNULL, stderr=subprocess.DEVNULL)
+    return send_file(a_out, mimetype="audio/wav")
+
+if __name__ == "__main__":
+    app.run(host="0.0.0.0", port=5002)

tts/requirements.txt

@@ -0,0 +1,2 @@
+flask==3.0.3
+piper-tts==1.2.0