核心代码目录结构完整代码python#!/usr/bin/python3 from asterisk.agi import * import os from websocket import create_connection import json import traceback AUDIO_FD 3 CONTENT_TYPE audio/l16; rate8000; channels1 ACCEPT audio/pcm def process_chunk(agi, ws, buf): agi.verbose(Processing chunk) ws.send_binary(buf) res json.loads(ws.recv()) agi.verbose(Result: str(res)) if result in res: text .join([w[word] for w in res[result]]) os.system(espeak -w /tmp/response22.wav \ text.encode(utf-8) \) os.system(sox /tmp/response22.wav -r 8000 /tmp/response.wav) agi.stream_file(/tmp/response) os.remove(/tmp/response.wav) def startAGI(): agi AGI() agi.verbose(EAGI script started...) ani agi.env[agi_callerid] did agi.env[agi_extension] agi.verbose(Call answered from: %s to %s % (ani, did)) ws create_connection(ws://localhost:2700) ws.send({ config : { sample_rate : 8000 } }) agi.verbose(Connection created) try: while True: data os.read(AUDIO_FD, 8000) if not data: break process_chunk(agi, ws, data) except Exception as err: agi.verbose(.join(traceback.format_exception(type(err), err, err.__traceback__)).replace(\n, )) finally: ws.close() startAGI()完整php#!/usr/bin/php -q ?php /* * author mihail.belobrovgmail.com * For FreePBX */ ob_implicit_flush(false); set_time_limit(0); error_reporting(0); $VersionPath dirname(__FILE__); $LibPath $VersionPath./lib; $voskServer ws://127.0.0.1:2700; $FD3 php://fd/3; require_once /etc/freepbx.conf; // Configure DB and other things require_once /var/lib/asterisk/agi-bin/phpagi.php; // Load php-agi in FreePBX $module_name eagiVosk; $dbh \FreePBX::Database(); // Connect to database $outGoingPath \FreePBX::Config()-get(ASTSPOOLDIR)./outgoing; // Recieve FreePBX settings $monitorPath \FreePBX::Config()-get(ASTSPOOLDIR)./monitor; // Call recordings storage // Keywords for call directions $word_arr array( tech ext-queues,8001,1, // queue 8001 accountant from-did-direct,102,1, // accountant 202 operator ext-queues,800,1, // quue 800 ); $myagi new AGI(); $myagi-answer(); // Loads websocket library require_once(/var/www/html/tts/vosk/scripts/websocket-php-1.3.1/vendor/autoload.php); use WebSocket\Client; $client new Client($voskServer, array(timeout 20,fragment_size 8192)); $client-send({ config : { sample_rate : 8000 } }, text); // Play hello sound $myagi-exec(PLAYBACK,array(/var/lib/asterisk/sounds/ru/hello-world)); $myText array(); $totalText ; $mydata fopen($FD3, rb); // opens file descriptor 3 $previous_text ; while(!feof($mydata)) { $data fread($mydata,8192); $client-send($data, binary); $receive $client-receive(); $result json_decode($receive, true); if(!empty($result[partial])) { $myagi-verbose($result[partial],3); if($previous_text ! $result[partial]){ $previous_text $result[partial]; foreach ($word_arr as $word $destination){ if (preg_match(/.$word./u, $previous_text , $matches) ){ list($context,$ext,$priority) explode(,,$destination); $myagi-exec_dial(Local,$ext..$context); // calls required direction break; } } } } } $client-send({\eof\ : 1}); $receive $client-receive(); fclose($mydata); $myagi-hangup(); ?Asterisk 插件Asterisk 插件独立维护在专属项目中https://github.com/alphacep/vosk-asterisk我们提供 Asterisk 模块以最大限度简化集成流程你只需编写如下拨号计划即可实现语音识别功能plaintext[internal] exten 1,1,Answer ; 接听来电 same n,Wait(1) ; 等待1秒 same n,SpeechCreate ; 创建语音识别会话 same n,SpeechBackground(hello) ; 播放hello语音并启动识别 same n,Verbose(0,Result was ${SPEECH_TEXT(0)}) ; 打印识别结果EAGI 方式你也可以将音频流转发至 AMI/ARI/AGI 接口由独立的 Web 应用处理音频数据。但从长期来看这种方式需要你基于 Statis 自行重构 Asterisk 的全部语音处理逻辑因此我们认为这并非实现语音交互界面的合理方案。从长远角度实现具备自然用户体验的语音输入功能最佳方案是采用异步处理机制。而异步输入处理需要比当前 Asterisk 语音 API 更复杂的技术架构支撑。未来我们可能会为 Asterisk 开发更完善的语音处理模块。你仍可参考本项目中的eagi.py文件体验 EAGI 功能使用方法如下安装依赖项asteriskAsterisk 核心程序docker容器运行环境TTS 语音合成包bash运行sudo apt install sox espeakAGI 及 WebSocket 开发包bash运行sudo pip3 install pyst2 websocket-client通过 Docker 启动识别服务器bash运行docker run -d -p 2700:2700 alphacep/kaldi-en:latest你也可选择自行部署 https://github.com/alphacep/vosk-server 并加载自定义模型。测试 EAGI 脚本示例脚本 eagi.py 可通过以下方式运行bash运行cd /home/user git clone https://github.com/alphacep/api-samples cd api-samples/asterisk python3 eagi.py ARGS: [eagi.py] # 脚本启动成功提示 ^C # 按CtrlC终止脚本配置 Asterisk 拨号计划编辑/etc/asterisk/extensions.conf原文笔误为 etc/extensions.confplaintextexten 200,1,Answer() ; 匹配200号码接听来电 same n,EAGI(/home/user/api-samples/asterisk/eagi.py) ; 调用EAGI脚本处理语音 same n,Hangup() ; 挂断通话测试通话并验证响应拨打 200 号码即可验证语音识别功能是否正常响应。如需实现更复杂的聊天机器人、数据库交互或呼叫流程定制可根据业务需求修改eagi.py。你也可使用 PHP/Perl 开发本项目中提供了对应示例。基于 PHP 的 EAGI 实现如需在 FreePBX原文笔误为 FreePBS环境下使用 PHP 实现 EAGI可参考示例脚本eagi_phpagi.php东方仙盟拥抱知识开源共筑数字新生态在全球化与数字化浪潮中东方仙盟始终秉持开放协作、知识共享的理念积极拥抱开源技术与开放标准。我们相信唯有打破技术壁垒、汇聚全球智慧才能真正推动行业的可持续发展。开源赋能中小商户通过将前端异常检测、跨系统数据互联等核心能力开源化东方仙盟为全球中小商户提供了低成本、高可靠的技术解决方案让更多商家能够平等享受数字转型的红利。共建行业标准我们积极参与国际技术社区与全球开发者、合作伙伴共同制定开放协议 与技术规范推动跨境零售、文旅、餐饮等多业态的系统互联互通构建更加公平、高效的数字生态。知识普惠共促发展通过开源社区 、技术文档与培训体系东方仙盟致力于将前沿技术转化为可落地的行业实践赋能全球合作伙伴共同培育创新人才推动数字经济 的普惠式增长阿雪技术观在科技发展浪潮中我们不妨积极投身技术共享。不满足于做受益者更要主动担当贡献者。无论是分享代码、撰写技术博客还是参与开源项目 维护改进每一个微小举动都可能蕴含推动技术进步的巨大能量。东方仙盟是汇聚力量的天地我们携手在此探索硅基 生命为科技进步添砖加瓦。Hey folks, in this wild tech - driven world, why not dive headfirst into the whole tech - sharing scene? Dont just be the one reaping all the benefits; step up and be a contributor too. Whether youre tossing out your code snippets , hammering out some tech blogs, or getting your hands dirty with maintaining and sprucing up open - source projects, every little thing you do might just end up being a massive force that pushes tech forward. And guess what? The Eastern FairyAlliance is this awesome place where we all come together. Were gonna team up and explore the whole silicon - based life thing, and in the process, well be fueling the growth of technology