pstn

智能电话外呼系统 呼叫中心系统 freeswitch webrtc

网关服务：Sip与Rtc互通，实现Web，Android，iOS，小程序，SIP座机，PSTN电话，手机互通。

Top geek's chindogu - personal assistant over satellite/telephone/SMS/chatbot, plus web infrastructure servers (web & mail, ad-free DNS, TCP-over-DNS, web proxy, etc)

园区访客登记 AI 语音 Agent：电话/浏览器/小程序三入口，对话采集车牌手机号 25 秒推企微访客卡 | voice agent, Twilio, WebRTC, FunASR, Qwen, FastAPI

How I replaced Skype with Twilio to make phone calls from my computer

☎️ A Matrix Puppet bridge for the public telephone network that supports a number of VoIP providers (Twillo, Vonage, etc.). Sends and receives voice and SMS.

This is the guide to show the method to build your own AI-Powered voice agent with LiveKit and Twillio

Skype for Business/Lync PSTN gateway

Code samples to Get started quickly with Symbl's Voice SDK and APIs: Node.js, JavaScript, WebSockets, & PSTN.

Make test calls to our PSTN phone numbers and SIP/VOIP addresses

Open-source, API-first voice CPaaS for browser + telecom calling. Drop in a <script> and a button - your agents place and receive real phone calls from the browser over FreeSWITCH. Multi-tenant, India-primary, Apache-2.0.

A sample app for enabling you to invite Symbl to your Zoom meetings by doing nothing more than simply copy and pasting the meeting invite in the app.

Python "Modem over IP" for Minitel

Microsoft Teams Observability Agent — call records, call queues & auto attendants telemetry. High quality and enriched metrics to Observability platforms.

Libary with Simple Temporal Networks (STN) for representing robot tasks

The Cryptofon Project aims to design a universal narrowband voice and text / data narrowband communication standard, that is encrypted and flexible to adapt for existing narrowband communication systems - aimed to be a more secure and open alternative to the propietary Secure Communications Interoperability Protocol (SCIP)

Microsoft Teams SIP PSTN gateway using Drachtio

🤖 Rox AI is a tool that connects Fonoster with Dialogflow ES/CX

ACL 2026 Demo: Real-time bidirectional Korean-English speech translation over PSTN telephone calls. Server-side relay with dual-session echo gating. 555ms median latency, USD 0.28/min.

PSTN / VoIP Configuration for Asterisk, HT-503, and Raspberry Pi