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Merged API samples into the main project
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‎README.md

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## Other programming languages
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See https://github.com/alphacep/api-samples
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Check other examples (Asterisk-EAGI, php, node, c#) in client-samples folder in this repository.

‎client-samples/asterisk/README-RU.md

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# Использование сервера распознавания с Asterisk
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### Устанавливаем зависимости
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- asterisk
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- docker
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- пакеты для работы со звуком:
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sudo apt install sox espeak
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- пакет для python AGI и вебсокетов
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sudo pip3 install pyst2 websocket-client
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### Запускаем сервер распознавания
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docker run -d -p 2700:2700 alphacep/kaldi-ru:latest
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### Проверяем, что можем запустить EAGI скрипт
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Код скрипта находится в файле [eagi-ru.py](https://github.com/alphacep/api-samples/blob/master/asterisk/eagi-ru.py) из этого пакета:
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```
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cd /home/user
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git clone https://github.com/alphacep/api-samples
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cd api-samples/asterisk
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python3 eagi-ru.py
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ARGS: ['eagi-ru.py']
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^C
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```
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### Настраиваем план звонков
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В etc/extensions.conf
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```
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exten => 200,1,Answer()
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same = n,EAGI(/home/user/api-samples/asterisk/eagi-ru.py)
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same = n,Hangup()
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```
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### Звоним и проверяем работу
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Для написания чатбота, сохранения результатов в базу, и так далее,
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изменяем код eagi-ru.py.
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‎client-samples/asterisk/README.md

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# Using kaldi-websocket-python server with Asterisk
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### Install dependencies
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- asterisk
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- docker
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- TTS packages:
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sudo apt install sox espeak
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- AGI and websocket package
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sudo pip3 install pyst2 websocket-client
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### Start recognition server with docker
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```
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docker run -d -p 2700:2700 alphacep/kaldi-en:latest
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```
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alternatively, you can run https://github.com/alphacep/kaldi-websocket-python with your models.
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### Test EAGI script
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Script [eagi.py](https://github.com/alphacep/api-samples/blob/master/asterisk/eagi.py) is located here:
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```
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cd /home/user
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git clone https://github.com/alphacep/api-samples
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cd api-samples/asterisk
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python3 eagi.py
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ARGS: ['eagi.py']
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^C
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```
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### Configure dialplan for Asterisk
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В etc/extensions.conf
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```
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exten => 200,1,Answer()
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same = n,EAGI(/home/user/api-samples/asterisk/eagi.py)
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same = n,Hangup()
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```
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### Call and check you get the response
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For more advanced chatbot, database interoperation and other callflow adjustments modify eagi.py according to your needs.
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You can also use PHP/Perl, see other examples in this package.

‎client-samples/asterisk/eagi-ru.py

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#!/usr/bin/python3
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from asterisk.agi import *
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import os
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from websocket import create_connection
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import json
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import uuid
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AUDIO_FD = 3
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CONTENT_TYPE = 'audio/l16; rate=8000; channels=1'
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ACCEPT = 'audio/pcm'
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def play_text(agi, text):
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fn = str(uuid.uuid4())
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os.system("espeak -v ru -w /tmp/%s.22.wav \"%s\"" % (fn, text))
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os.system("sox /tmp/%s.22.wav -r 8000 /tmp/%s.wav" % (fn, fn))
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agi.stream_file("/tmp/%s" % (fn))
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os.remove("/tmp/%s.22.wav" % (fn))
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os.remove("/tmp/%s.wav" % (fn))
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os.read(AUDIO_FD, 1000000) # Read remaining chunks
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def process_chunk(agi, ws, buf):
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ws.send_binary(buf)
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res = json.loads(ws.recv())
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agi.verbose(str(res))
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if 'text' in res:
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play_text(agi, "Распознано " + res['text'])
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def startAGI():
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agi = AGI()
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agi.verbose("EAGI script started...")
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ani = agi.env['agi_callerid']
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did = agi.env['agi_extension']
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agi.verbose("Call answered from: %s to %s" % (ani, did))
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play_text(agi, "Привет")
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ws = create_connection("ws://localhost:2700")
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try:
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while True:
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data = os.read(AUDIO_FD, 8000)
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if not data:
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break
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process_chunk(agi, ws, data)
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finally:
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ws.close()
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startAGI()

‎client-samples/asterisk/eagi.py

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#!/usr/bin/python3
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from asterisk.agi import *
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import os
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from websocket import create_connection
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import json
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AUDIO_FD = 3
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CONTENT_TYPE = 'audio/l16; rate=8000; channels=1'
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ACCEPT = 'audio/pcm'
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def process_chunk(agi, ws, buf):
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ws.send_binary(buf)
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res = json.loads(ws.recv())
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if 'result' in res:
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text = " ".join([w['word'] for w in res['result']])
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agi.verbose(text)
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os.system("espeak -w /tmp/response22.wav \"" + text.encode('utf-8') + "\"")
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os.system("sox /tmp/response22.wav -r 8000 /tmp/response.wav")
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agi.stream_file("/tmp/response")
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os.remove("/tmp/response.wav")
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def startAGI():
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agi = AGI()
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agi.verbose("EAGI script started...")
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ani = agi.env['agi_callerid']
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did = agi.env['agi_extension']
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agi.verbose("Call answered from: %s to %s" % (ani, did))
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ws = create_connection("wss://api.alphacephei.com/asr/en")
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try:
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while True:
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data = os.read(AUDIO_FD, 8000)
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if not data:
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break
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process_chunk(agi, ws, data)
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finally:
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ws.close()
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startAGI()

‎client-samples/csharp/test.cs

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// Requires at least Mono 5.0 due to TLS issues
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using System;
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using System.IO;
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using System.Text;
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using System.Threading;
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using System.Threading.Tasks;
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using System.Net.WebSockets;
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public class ASRTest
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{
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async Task RecieveResult(ClientWebSocket ws) {
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byte[] result = new byte[4096];
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Task<WebSocketReceiveResult> receiveTask = ws.ReceiveAsync(new ArraySegment<byte>(result), CancellationToken.None);
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await receiveTask;
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var receivedString = Encoding.UTF8.GetString(result, 0, receiveTask.Result.Count);
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Console.WriteLine("Result {0}", receivedString);
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}
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async Task ProcessData(ClientWebSocket ws, byte[] data, int count) {
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await ws.SendAsync(new ArraySegment<byte>(data, 0, count), WebSocketMessageType.Binary, true, CancellationToken.None);
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await RecieveResult(ws);
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}
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async Task ProcessFinalData(ClientWebSocket ws) {
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byte[] eof = Encoding.UTF8.GetBytes("{\"eof\" : 1}");
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await ws.SendAsync(new ArraySegment<byte>(eof), WebSocketMessageType.Text, true, CancellationToken.None);
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await RecieveResult(ws);
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}
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async Task DecodeFile() {
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ClientWebSocket ws = new ClientWebSocket();
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await ws.ConnectAsync(new Uri("wss://api.alphacephei.com/asr/en/"), CancellationToken.None);
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FileStream fsSource = new FileStream("test.wav",
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FileMode.Open, FileAccess.Read);
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byte[] data = new byte[8000];
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while (true) {
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int count = fsSource.Read(data, 0, 8000);
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if (count == 0)
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break;
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await ProcessData(ws, data, count);
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}
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await ProcessFinalData(ws);
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await ws.CloseAsync(WebSocketCloseStatus.NormalClosure, "OK", CancellationToken.None);
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}
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public static void Main() {
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Task.Run(async () => {
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await new ASRTest().DecodeFile();
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}).GetAwaiter().GetResult();
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}
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}

‎client-samples/node/test.js

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const websocket = require('ws');
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const fs = require('fs');
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const ws = new websocket('wss://api.alphacephei.com/asr/en/');
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ws.on('open', function open() {
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var readStream = fs.createReadStream('test.wav');
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readStream.on('data', function (chunk) {
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ws.send(chunk);
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});
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readStream.on('end', function () {
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ws.send('{"eof" : 1}');
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});
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});
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ws.on('message', function incoming(data) {
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console.log(data);
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});
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ws.on('close', function close() {
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process.exit()
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});

‎client-samples/php/asr-test.php

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<?php
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require_once("./vendor/autoload.php");
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use WebSocket\Client;
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$client = new Client("wss://api.alphacephei.com/asr/en/", array('timeout' => 200));
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$myfile = fopen("test.wav", "r");
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while(!feof($myfile)) {
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$data = fread($myfile, 8000);
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$client->send($data, 'binary');
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echo $client->receive() . "\n";
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}
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$client->send("{\"eof\" : 1}");
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echo $client->receive() . "\n";
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fclose($myfile);
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?>

‎client-samples/php/composer.json

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{
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"require": {
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"textalk/websocket": "1.0.*",
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"marcelog/pagi": "2.*"
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}
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}
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‎client-samples/php/tts-test.php

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<?php
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require_once("./vendor/autoload.php");
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$url = 'https://api.alphacephei.com/tts';
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$data = array('text' => 'Hello world!');
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$options = array(
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'http' => array(
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'header' => "Content-type: application/x-www-form-urlencoded\r\n",
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'method' => 'POST',
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'content' => http_build_query($data)
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)
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);
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$context = stream_context_create($options);
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$result = file_get_contents($url, false, $context);
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if ($result === FALSE) { /* Handle error */ }
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file_put_contents("tts.wav", $result);
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?>

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