Setup for a WEBRTC client and Kamailio server to call SIP clients

Official one-stop shop for AI Agents and developers building with Telnyx.

Barebone Opensource Powered SBC

P-KISS-SBC - simple and stupid SIP/RTP SBC - AGPL v3 - Based on kamailio / RTP Engine

Ansible scripts to install and configure SIP3

Dockerized FreePBX 17 with Asterisk 21, PHP 8,2, dedicated MariaDB and data persistence

FreePBX container (Asterisk 16; OpenPBX 15 with Backup and IVR modules installed)

Dockerfile for freeswitch

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

openduplex uses speech-to-text, artificial intelligence and text-to-speech, to call businesses and make appointments for you

OpenIMSs open source environment for real life development of IMS based 4G/5G/NR voice/video/data/RCS/IM services

How To Use Router as Sip Server (Asterisk openwrt) ...

The callflow sequence diagram generator is a collection of awk and shell scripts that will take a packet capture file that can be read by wireshark and produce a time sequence diagram. This is useful to view & debug SIP callflows or other network traffic.

Kamailio in docker container with TLS enabled using Let's Encrypt.

Skype for Business/Lync PSTN gateway

Jami (formerly Ring (formerly SFLphone)) is a SIP compatible softphone for GNU/Linux.

Docker container for running Kamailio as part of a WebRTC Gateway

🐋 Minimal FreeSWITCH Docker images

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

Simple scripts for 3CX server installation and management