Open Source Mac Telephony Software
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Telephony Software for Mac
View 102 business solutionsBrowse free open source Telephony software and projects for Mac below. Use the toggles on the left to filter open source Telephony software by OS, license, language, programming language, and project status.
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IssabelPBX
Issabel PBX - Unified Communications
Open Source and Unified Communications partners created a new platform based on an Elastix® fork (currently purchased by 3CX) to provide the community with continuity, peace of mind and support needed to continue with their PBX and operation developments. Contribute to the funding of Issabel on https://www.patreon.com/issabel -
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NoiseGator (Noise Gate)
A simple noise gate app intended for use with VOIPs like Skype.
Ever wanted to cut out background noise when talking with others on Skype? Now it's possible! NoiseGator is a light-weight noise gate application that routes audio through an audio input to an audio output. In real-time the audio level is analysed and if the average level is higher than the threshold the audio bypasses as normal. However, if the average level goes below the threshold, the gate closes and the audio is cut. When used with a virtual audio cable it can act as a noise gate for a either a sound input(microphone) or sound output(speakers). Can also be used to gate noise from your own mic or play your microphone through your speakers. REQUIREMENTS: - Java 7 or higher for Windows. - Java 6 or higher for Mac. Java 7 recommended. - A virtual audio cable is required for use with VOIPs: For Windows users I recommend the VB-Cable driver (http://vb-audio.pagesperso-orange.fr/Cable/index.htm). Mac users can use SoundFlower. -
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Elastix
Unified Communications Server
Elastix is a software-based PBX powered by 3CX and based on Debian. An open-standards solution, Elastix is an easy to install and manage UC system compatible with popular IP phones, gateways and SIP trunks. Elastix is complete with unified communications features such as integrated WebRTC video conferencing, chat, presence and softphones and smartphone clients for Windows, Mac, iOS and Android. -
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trixbox CE is an easy to install, VOIP phone system based on the Asterisk PBX. trixbox is designed for home or office use. trixbox CE includes CentOS linux, mysql, and all the tools needed to run a business quality phone system. (formerly asterisk@home)
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Asterisk GUI client, VICIdial
VICIdial Contact Center Suite
This software suite is designed to extend the functionality of the Asterisk PBX through platform-independant web-client applications. Includes the VICIdial inbound/outbound contact center application. The suite is scalable across multiple Asterisk servers. -
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This project implements a simple STUN server and client on Windows, Linux, and Solaris. The STUN protocol (Simple Traversal of UDP through NATs) is described in the IETF RFC 3489, available at http://www.ietf.org/rfc/rfc3489.txt
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BlackBelt WASTE - ipv4/Tor/i2p +AI+Voice
Modern, AI-Smart, WASTE p2p for ipv4, Tor and i2p + Voice Conference.
Open Source - GPLv3 inc images. A WASTE client. Download and create your own WASTE networks. Move 1000's of GB's at 100MB+ per sec. (800 Mbits per sec) FULL pause and resume capable. Voice Conference, Chat, Transfer files and Participate in Forums in a secure environment. For Windows XP 32/64, Vista 32/64, Win7 32/64, Win8 32/64, Win 10, Win 11, Linux (WINE). *** User Based Access Control - for voice, chats, file transfers and uploads. (useful in NULLNETS) *** Distributed Autonomic-Performance-Tuning - A Goal-Seeking Swarming-Semiotic AI *** AI Connect - AI Managed Connections. *** Self-Organising Anti-Spoofing Technology *** Geared Multi-threading, providing the smoothest performance possible *** Advanced Threat Detect and Manage Technology *** Voice Conferencing Over WASTE *** RNN - Recurring Neural Net - AI Noise Reduction *** Differential Files Transfer - Seriously fast data backups -
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oreka
Enterprise telephony recording and retrieval system
Enterprise telephony recording and retrieval system with web based user interface. The project currently supports recording voice from VoIP SIP, Cisco Skinny (aka SCCP), raw RTP and audio sound device and runs on multiple operating systems and database systems. It can record audio from most PBX and telephony systems such as BroadWorks, Metaswitch, Asterisk, FreeSwitch, OpenSIPS, Avaya, Nortel, Mitel, Siemens, Cisco Call Manager, Cosmocom, NEC, etc... It is amongst others being used in Call Centers and Contact Centers for Quality monitoring (QM) purposes. -
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JDA
Java wrapper for the popular chat & VOIP service
JDA strives to provide a clean and full wrapping of the Discord REST api and its Websocket-Events for Java. This library is a helpful tool that provides the functionality to create a discord bot in java. Discord is currently prohibiting the creation and usage of automated client accounts (AccountType.CLIENT). We have officially dropped support for client login as of version 4.2.0! Note that JDA is not a good tool to build a custom discord client as it loads all servers/guilds on startup, unlike a client which does this via lazy loading instead. If you need a bot, use a bot account from the Application Dashboard. Creating the JDA Object is done via the JDABuilder class. After setting the token and other options via setters, the JDA Object is then created by calling the build() method. When build() returns, JDA might not have finished starting up. However, you can use await ready() on the JDA object to ensure that the entire cache is loaded before proceeding. -
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Siproxd is a proxy/masquerading daemon for the SIP protocol. It allows SIP clients (softphones & hardphones) to work behind an IP masquerading firewall or router.
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Longing for the good old days of Asterisk@Home? Welcome back to the steroid-enhanced version. PBX in a Flash 3.0 & Incredible PBX 2020/2021/2022/2027 are the latest Lean, Mean Asterisk Machines, high-performance, turnkey Asterisk PBXs that are easy to upgrade. Features include Rocky8, CentOS/SL 7.x, Ubuntu 22.04 & 20.04, Debian 10 and Raspbian 10 support with Asterisk 20/18/16 and FreePBX 16/15 GPL modules. Add-ons include one-click installs of Incredible Fax and many other Asterisk utilities. Visit Nerd Vittles for the latest tutorials.
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Open Phone Abstraction Library (OPAL) is a C++ multi-platform, multi-protocol library for Fax, Video & Voice over IP and other networks. Also included is the Portable Tool Library (PTLib) which is a C++ multi-platform abstraction library and collection o
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What is t38modem? From your application view point it's a fax/voice modem pool. From IP network view point it's a H.323/SIP endpoint with T.38 fax support. From your view point it's a gateway between an application and IP network. Works with HylaFAX.
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Gammu
Cellular manager for mobile phones/modems
Gammu is a cellular manager for mobile phones/modems. It contains libraries and functions for ringtones,logos,phonebook,SMS,etc. (used by external software), a command line version (with backup/restore) and SMS gateway (with MySQL and PostgreSQL supp -
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SIM Explorer
Sim Card explorer and decoder
This software uses your smart card reader/terminal to navigate through the SIM Card directory tree. The contents of the files are decoded and showed to the user. Other features, such as CHV management and SIM backup, are also available. -
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The NCID (Network Caller ID) project is Caller ID (CID) distributed over a network. The project contains the NCID package and 4 optional client packages. Each package is described at the NCID web site. A non-inclusive list of 3rd party addons is also available at the web site Available Packages: NCID - contains the server, gateways, and a client with output modules LCDncid - a client that uses LCDproc to display Caller ID on a LCD display NCIDandroid - a client and gateway for Android devices NCIDdisplay - a homebrew client that displays on large LED modules NCIDpop - a popup client for Windows, Mac, and Linux
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atinout
AT commands as input are sent to modem and responses given as output.
This program will read a file (or stdin) containing a list of AT commands. Each command will be send to the modem, and all the response for the command will be output to file (or stdout). Example, to hang up any ongoing call: $ echo ATH | atinout - /dev/ttyACM0 - ATH OK $ -
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JFritz is a application written in Java, which makes it possible for AVM FRITZ!Box users to download the caller history from their box and manage it on their desktop pc.
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008
Open-source event-driven AI powered Softphone
008 is an open-source event-driven AI powered WebRTC Softphone compatible with macOS, Windows, and Linux. It is also accessible on the web (though official support for browser-related issues is not provided). The name '008' or 'agent 008' reflects our ambition: beyond crafting the premier Open Source Softphone, we aim to introduce a programmable, event-driven AI agent. This agent utilizes embedded artificial intelligence models operating directly on the softphone, ensuring efficiency and reduced operational costs. This project is a WebRTC softphone, and communication is achieved via SIP over a socket. Leading PBX systems like Asterisk or Freeswitch support socket connections. If your provider does not offer this feature, consider using a SIP proxy such as Kamailio, Opensip or Routr. The softphone is internally configured using a JSON definition. The configuration file can be loaded from either a server or a local file. 008 reads the file only once. -
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Amazon Connect connect-rtc-js
Provide softphone support to AmazonConnect customers
connect-rtc.js provides softphone support to AmazonConnect customers when they choose to directly integrate with AmazonConnect API and not use the AmazonConnect web application. It implements Amazon Connect WebRTC signaling protocol and integrates with browser WebRTC APIs to provide a simple contact session interface that can seamlessly integrate with Amazon Connect StreamJS. In a typical amazon-connect-streams integration, connect-rtc-js is not required on parent page. Softphone call handling is done by embedded CCP. Load connect-rtc-js along with amazon-connect-streams on parent page. In the gh-pages branch prebuilt ready-to-use files can be downloaded/linked directly. -
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Twisted protocols providing access to the Asterisk PBX's AMI and FastAGI interfaces from within the same programming process.
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This project has been superseded by OpalVoip (https://sourceforge.net/projects/opalvoip/) and H323Plus (https://sourceforge.net/projects/h323plus/) The OpenH323 project provides full featured, interoperable, Open Source implementation of the ITU H.323 teleconferencing protocol that can be used by personal developers and commercial users without charge.
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baresip
Baresip is a modular SIP User-Agent with audio and video support
Baresip is a portable and modular SIP User-Agent with audio and video support. the latest source code can be found here: https://github.com/alfredh/baresip -
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Stuntman - STUN server and client
High performance, production quality STUN server and client library
New version 1.2. This is the code to STUNTMAN - an open source STUN server and client code by john selbie. Compliant with the latest RFCs including 5389, 5769, and 5780. Also includes backwards compatibility for RFC 3489. ICE and WebRTC ready. Version 1.2 compiles on Linux, MacOS, BSD, and Solaris. Supports the STUN protocol on both UDP and TCP for both IPv4 and IPv6. Windows binaries are also provided. Additional features are in development. This is a mirror of the code on https://github.com/jselbie/stunserver More details on the project's website: http://www.stunprotocol.org -
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