Open Source Linux Telephony Software
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Telephony Software for Linux
View 48 business solutionsBrowse free open source Telephony software and projects for Linux 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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VoIP monitor
VoIP SIP and SKINNY quality analyzer and packet / audio recording tool
VoIPmonitor is open source network packet sniffer with commercial frontend for SIP SKINNY MGCP RTP and RTCP VoIP protocols running on linux. VoIPmonitor is designed to analyze quality of VoIP call based on network parameters - delay variation and packet loss according to ITU-T G.107 E-model which predicts quality on MOS scale. Calls with all relevant statistics are saved to MySQL or ODBC database. Optionally each call can be saved to pcap file with either only SIP / SKINNY protocol or SIP/RTP/RTCP/T.38/udptl protocols. VoIPmonitor can also decode audio. -
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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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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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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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Sipp is a performance testing tool for the SIP protocol. Its main features are basic SIPStone scenarios, TCP/UDP transport, customizable (XML-based) scenarios, dynamic adjustment of call-rate and a comprehensive set of real-time statistics.
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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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FusionPBX
A domain based multi-tenant PBX & voice switch for FreeSWITCH
FusionPBX is a multi-tenant PBX and voice switch for FreeSWITCH, a highly scalable, multi-threaded, cross-platform communication platform. FusionPBX can be used either as a single or domain based PBX, and offers several functionalities: as carrier grade switch, call center server, voicemail server, fax server, conference server, VoIP server, multi-tenant appliance framework and so much more. It also offers unlimited extensions, device provisioning, music on hold, call parking, automatic call distribution and many other superb features. Give your company the business functionality and corporate level phone system it needs with FusionPBX! -
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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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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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Synapse
Matrix reference homeserver
Matrix is an ambitious new ecosystem for open federated Instant Messaging and VoIP. Everything in Matrix happens in a room. Rooms are distributed and do not exist on any single server. Rooms can be located using convenience aliases like #matrix:matrix.org or #test:localhost:8448. Synapse is currently in rapid development, but as of version 0.5 we believe it is sufficiently stable to be run as an internet-facing service for real usage! Create and manage fully distributed chat rooms with no single points of control or failure. Eventually-consistent cryptographically secure synchronization of room state across a global open network of federated servers and services. Send and receive extensible messages in a room with (optional) end-to-end encryption. Use 3rd Party IDs (3PIDs) such as email addresses, phone numbers, Facebook accounts to authenticate, identify and discover users on Matrix. -
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Seagull is a multi-protocol traffic generator. Especially targeted towards IMS, Seagull supports Diameter (RFC3588 and all applications) over TCP/SCTP and IPv4/IPv6 , TCAP (over SS7 or Sigtran), XCAP over HTTP and Radius.
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libphonenumber for PHP
PHP version of Google's phone number handling library
A PHP library for parsing, formatting, storing and validating international phone numbers. This library is based on Google's libphonenumber. This library will try to follow the same version numbers as Google. There could be additional releases where needed to fix critical issues that can not wait until the next release from Google. This does mean that this project may not follow Semantic Versioning, but instead Google's version policy. As a result, jumps in major versions may not actually contain any backward-incompatible changes. Please read the release notes for such releases. This library uses phone number metadata from Google's libphonenumber. If this library is working as intended, it should provide the same result as the Java version of Google's project. If you believe that a phone number is returning an incorrect result, first test it with libphonenumber via their Online Demo. -
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IAXmodem is a software modem written in C that uses an IAX channel (commonly provided by an Asterisk PBX system) instead of a traditional phone line and uses a DSP library instead of DSP hardware chipsets.
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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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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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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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YetiForce CRM
The most affordable CRM system according to Capterra
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anyremote
Remote control software for applications using wi-fi or bluetooth
Remote control software for applications using wi-fi or bluetooth. -
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Kiax is a softphone (soft phone, VoIP client) with a simple and comfortable user interface for making VoIP calls to Asterisk PBX. It depends on the iaxclient library to use Asterisk's IAX2 protocol for easy call configuration and audio setup.
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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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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
