Asterisk の音声データを Incoming Webhook で MiiTel に連携する

# リポジトリの更新
apt-get update

# 必要なパッケージのインストール
apt-get install bison wget openssl libssl-dev libasound2-dev libc6-dev \
  libxml2-dev libsqlite3-dev libnewt-dev libncurses5-dev zlib1g-dev \
  gcc g++ make perl uuid-dev git subversion libjansson-dev \
  unixodbc-dev unixodbc autoconf libedit-dev bzip2 pkg-config

cd /usr/src

wget http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/asterisk-<version>.tar.gz
tar -xzvf asterisk-<version>.tar.gz

# ビルドとインストール
cd asterisk-<version>
./configure
make && make install && make config && make samples

sudo systemctl enable asterisk
sudo systemctl start asterisk
asterisk -r

mv /etc/asterisk/pjsip.conf /etc/asterisk/pjsip.conf.bak
vi /etc/asterisk/pjsip.conf

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0:5060
external_media_address=<YOUR_PUBLIC_IP>
external_signaling_address=<YOUR_PUBLIC_IP>
local_net=<YOUR_PRIVATE_SUBNET>/255.255.255.0

;================================================================================
; SIP クライアント
;================================================================================
[6000]
type=endpoint
transport=transport-udp
context=from-internal
disallow=all
allow=ulaw
aors=6000
auth=6000

[6000]
type=auth
auth_type=userpass
username=6000
password=<YOUR_PASSWORD>

[6000]
type=aor
max_contacts=2

;================================================================================
; SIP トランクテンプレート (Twilio の例 — プロバイダーに合わせて調整してください)
;================================================================================
[twilio-endpoint](!)
type=endpoint
transport=transport-udp
context=from-siptrunk
disallow=all
allow=ulaw
force_rport=yes
rtp_symmetric=yes
rtp_keepalive=10
rtp_timeout=30
rtp_timeout_hold=60
from_domain=<YOUR_TWILIO_PSTN>.pstn.twilio.com

[twilio-identify](!)
type=identify
endpoint=twilio-trunk
; お使いのリージョンの Twilio IP 範囲を追加してください。
; 参照: https://www.twilio.com/docs/sip-trunking/ip-addresses
match=<TWILIO_IP_RANGE>

;================================================================================
; 電話番号ごとの SIP トランク (Twilio の例 — プロバイダーに合わせて調整してください)
;================================================================================
[twilio<YOUR_PHONE_NUMBER>](twilio-endpoint)
aors=twilio<YOUR_PHONE_NUMBER>
outbound_auth=twilio<YOUR_PHONE_NUMBER>

[twilio<YOUR_PHONE_NUMBER>]
type=auth
auth_type=userpass
username=twilio_user
password=<YOUR_PASSWORD>

[twilio<YOUR_PHONE_NUMBER>]
type=aor
max_contacts=10
contact=sip:<YOUR_TWILIO_PSTN>.pstn.twilio.com

[twilio<YOUR_PHONE_NUMBER>](twilio-identify)
endpoint=twilio<YOUR_PHONE_NUMBER>

mv /etc/asterisk/extensions.conf /etc/asterisk/extensions.conf.bak
vi /etc/asterisk/extensions.conf

[globals]
COUNTRY_CODE=81
CALLERID_MAP_6000=<YOUR_PHONE_NUMBER>

[from-internal]
; 外部への発信
exten => _[0-9+].,1,Noop(from-internal)
 same =>   n,Ringing()
 same =>   n,Progress()
 same =>   n,Set(VOLUME(TX)=1)
 same =>   n,Set(VOLUME(RX)=1)
 same =>   n,Set(E164_NUMBER=${EXTEN})
 same =>   n,ExecIf($["${E164_NUMBER:0:1}"="+"]?Set(E164_NUMBER=${E164_NUMBER}):Set(E164_NUMBER=+${GLOBAL(COUNTRY_CODE)}${E164_NUMBER}))
 same =>   n,ExecIf($["${E164_NUMBER:0:1}"="0"]?Set(E164_NUMBER=+${GLOBAL(COUNTRY_CODE)}${E164_NUMBER:1}):Set(E164_NUMBER=${E164_NUMBER}))
 same =>   n,Set(CALLERID_NUM=${CALLERID_MAP_${CALLERID(num)}})
 same =>   n,Set(CALLERID(name)=+${CALLERID_NUM})
 same =>   n,Set(CALLERID(num)=+${CALLERID_NUM})
 same =>   n,Set(RECORDING_UUID=${UUID()})
 same =>   n,MixMonitor(${RECORDING_UUID}.raw,D)
 same =>   n,Set(CHANNEL(hangup_handler_push)=hangup_handler,s,1(${RECORDING_UUID},${CALLERID(num)},${E164_NUMBER},OUTGOING_CALL))
 same =>   n,Dial(PJSIP/${E164_NUMBER}@twilio${CALLERID_NUM})
 same =>   n,Return()

; 外部からの着信
[from-siptrunk]
exten => +<YOUR_PHONE_NUMBER>,1,GoSub(dial-extension,s,1(6000,${EXTEN}))

[dial-extension]
exten => s,1,Noop(from-siptrunk)
 same =>   n,Set(VOLUME(TX)=1)
 same =>   n,Set(VOLUME(RX)=1)
 same =>   n,Set(RECORDING_UUID=${UUID()})
 same =>   n,MixMonitor(${RECORDING_UUID}.raw,D)
 same =>   n,Set(CHANNEL(hangup_handler_push)=hangup_handler,s,1(${RECORDING_UUID},${CALLERID(num)},${ARG2},INCOMING_CALL))
 same =>   n,Dial(PJSIP/${ARG1})
 same =>   n,Return()

[hangup_handler]
exten => s,1,NoOp(Hangup handler: UUID=${ARG1} FROM=${ARG2} TO=${ARG3} DIR=${ARG4})
 same =>   n,AGI(call_webhooks.py,${ARG1},${ARG2},${ARG3},${ARG4})
 same =>   n,Return()

same => n,MixMonitor(${RECORDING_UUID}.raw,D)

import wave

with open(raw_file, "rb") as f:
    raw_data = f.read()
with wave.open(wav_file, "wb") as w:
    w.setnchannels(2)       # ステレオ
    w.setsampwidth(2)       # 16-bit
    w.setframerate(8000)    # 8kHz (電話音声の標準)
    w.writeframes(raw_data)

same => n,MixMonitor(${RECORDING_UUID}.wav,r(${RECORDING_UUID}-in.wav)t(${RECORDING_UUID}-out.wav))

sox -M ${RECORDING_UUID}-in.wav ${RECORDING_UUID}-out.wav ${RECORDING_UUID}-stereo.wav

ls -la /var/spool/asterisk/monitor/

vi /var/lib/asterisk/agi-bin/call_webhooks.py

#!/usr/bin/python3

import sys
import os
import time
import wave
import requests
from datetime import datetime, timezone

# === Read AGI environment ===
agi_env = {}
while True:
    line = sys.stdin.readline().strip()
    if not line:
        break
    if ': ' in line:
        key, _, value = line.partition(': ')
        agi_env[key] = value

def agi_command(cmd):
    sys.stdout.write(f"{cmd}\n")
    sys.stdout.flush()
    return sys.stdin.readline().strip()

def get_variable(name):
    result = agi_command(f"GET VARIABLE {name}")
    if "1 (" in result:
        return result.split("(", 1)[1].rstrip(")")
    return ""

def to_iso(asterisk_time):
    if not asterisk_time:
        return None
    try:
        dt = datetime.strptime(asterisk_time, "%Y-%m-%d %H:%M:%S")
        return dt.replace(tzinfo=timezone.utc).strftime("%Y-%m-%dT%H:%M:%SZ")
    except ValueError:
        return None

# === AGI arguments from dialplan ===
RECORDING_UUID = agi_env.get('agi_arg_1', '')
FROM_NUMBER = agi_env.get('agi_arg_2', '')
TO_NUMBER = agi_env.get('agi_arg_3', '')
DIRECTION = agi_env.get('agi_arg_4', 'OUTGOING_CALL')

# === Get call timestamps from CDR ===
call_starts_at = to_iso(get_variable("CDR(start)"))
call_answered_at = to_iso(get_variable("CDR(answer)"))
call_ends_at = to_iso(get_variable("CDR(end)"))

if not call_ends_at:
    call_ends_at = datetime.now(timezone.utc).strftime("%Y-%m-%dT%H:%M:%SZ")
if not call_starts_at:
    call_starts_at = call_ends_at

# === Config ===
TENANT_CODE = "<YOUR_TENANT_CODE>"
WEBHOOK_ID = "<YOUR_WEBHOOK_ID>"
RECORDING_DIR = "/var/spool/asterisk/monitor"
BASE_URL = f"https://{TENANT_CODE}.miitel.jp/api/call/v2/webhook_pbx/{WEBHOOK_ID}"
YOUR_MIITEL_USER_UUID = "<YOUR_MIITEL_USER_UUID>"

RAW_FILE = f"{RECORDING_DIR}/{RECORDING_UUID}.raw"
WAV_FILE = f"{RECORDING_DIR}/{RECORDING_UUID}-stereo.wav"

# === Convert raw interleaved stereo to wav ===
agi_command(f'VERBOSE "Converting {RAW_FILE} to stereo wav" 1')

if not os.path.exists(RAW_FILE):
    agi_command(f'VERBOSE "Raw file not found: {RAW_FILE}" 1')
    sys.exit(0)

try:
    with open(RAW_FILE, "rb") as raw:
        raw_data = raw.read()
    with wave.open(WAV_FILE, "wb") as w:
        w.setnchannels(2)
        w.setsampwidth(2)
        w.setframerate(8000)
        w.writeframes(raw_data)
except Exception as e:
    agi_command(f'VERBOSE "Wav conversion failed: {e}" 1')
    sys.exit(0)

# === Build participants ===
if DIRECTION == "OUTGOING_CALL":
    from_user_id = YOUR_MIITEL_USER_UUID
    to_user_id = None
else:
    from_user_id = None
    to_user_id = YOUR_MIITEL_USER_UUID

# === Step 1: POST call metadata ===
payload = {
    "call_data": [
        {
            "call_data_id": f"call-{int(time.time())}",
            "audio_file_type": "wav",
            "audio_channel_type": "STEREO",
            "event_type": DIRECTION,
            "call_starts_at": call_starts_at,
            "call_answered_at": call_answered_at,
            "call_ends_at": call_ends_at,
            "circuit_number": FROM_NUMBER,
            "group_name": None,
            "queue_name": None,
            "participants": [
                {
                    "type": "FROM",
                    "stereo_lr": "left",
                    "miitel_user_id": from_user_id,
                    "number": FROM_NUMBER,
                    "name": "Agent" if DIRECTION == "OUTGOING_CALL" else "Customer",
                },
                {
                    "type": "TO",
                    "stereo_lr": "right",
                    "miitel_user_id": to_user_id,
                    "number": TO_NUMBER,
                    "name": "Customer" if DIRECTION == "OUTGOING_CALL" else "Agent",
                },
            ],
            "tags": [{"value": "Record from asterisk"}],
        }
    ]
}

agi_command(f'VERBOSE "Webhook: {DIRECTION} from={FROM_NUMBER} to={TO_NUMBER}" 1')

resp = requests.post(BASE_URL, json=payload, timeout=30)
agi_command(f'VERBOSE "Webhook status: {resp.status_code}" 1')

if resp.status_code != 201:
    agi_command(f'VERBOSE "Webhook failed: {resp.text}" 1')
    sys.exit(0)

data = resp.json()
upload_info = list(data["audio_upload_urls"][0].values())[0]
presigned_url = upload_info["url"]

# === Step 2: Upload stereo .wav ===
agi_command(f'VERBOSE "Uploading {WAV_FILE}" 1')

with open(WAV_FILE, "rb") as f:
    upload_resp = requests.put(presigned_url, data=f, timeout=60)

agi_command(f'VERBOSE "Upload status: {upload_resp.status_code}" 1')

# === Cleanup raw file ===
try:
    os.remove(RAW_FILE)
except OSError:
    pass

sys.exit(0)

chmod +x /var/lib/asterisk/agi-bin/call_webhooks.py

sudo asterisk -vvvvvr