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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Fully Managed MySQL, PostgreSQL, and SQL ServerCloud SQL handles your database ops end to end, so you can focus on your app.
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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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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. -
Enterprise-grade ITSM, for every businessFreshservice is an intuitive, AI-powered platform that helps IT, operations, and business teams deliver exceptional service without the usual complexity. Automate repetitive tasks, resolve issues faster, and provide seamless support across the organization. From managing incidents and assets to driving smarter decisions, Freshservice makes it easy to stay efficient and scale with confidence.
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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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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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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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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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Add Two Lines of Code. Get Full APM.Works out of the box for Rails, Django, Express, Phoenix, and more. Monitoring exceptions and performance in no time.
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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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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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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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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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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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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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YetiForce CRM
The most affordable CRM system according to Capterra
Are you looking for a system to boost your sales? Do you want to improve your relationships with customers? Do you want to contribute to the fastest growing open source project on GitHub? You can use YetiForce completely free of charge. With YetiForce’s help you will manage 12 business processes in your company, no matter the size. The vast number of features available for free helped us win the first place in Capterra’s “Most Affordable CRM Software” ranking. YetiForce’s open source code was released under a very flexible and liberal license, which gathers a large and active community around the project. You too can contribute to it no matter if you are a professional programmer or software development is just your hobby. Download YetiForce for free and find out why you should become one of more than 250 000 people who already use it. Testing versions: GitStable: https://gitstable.yetiforce.com/ GitDeveloper: https://gitdeveloper.yetiforce.com/ -
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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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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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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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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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jfBroadcast
VoIP/SIP Autodialer
VoIP/SIP AutoDialer. Broadcasts a message with the option to transfer person to another number. Now includes new survey options. -
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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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The ASN.1 Compiler
Go to github.com/vlm/asn1c for the latest version.
This ASN.1 compiler turns ASN.1 specifications into C code. The asn1c is shipped together with conformant BER/DER/XER/PER codecs. The X.509, GSM TAP3, MEGACO, RRC and LDAP encoding and decoding examples are part of the source code distribution. NOTE: THE asn1c PROJECT HAS LARGELY MOVED TO GITHUB: http://github.com/vlm/asn1c -
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PHPAGI is a PHP Class for writing AGI applications for use with the open source Asterisk PBX software.
