Open Source Telephony Software
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Telephony Software
View 312 business solutionsBrowse free open source Telephony software and projects 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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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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Nokia NBU, NBF, NFB, NFC and ARC backup file parser, extractor and viewer. It can help you to check content of backup or extract files from it. Requires MS .Net Framework 2
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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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PhoneNumberKit
A Swift framework for parsing and formatting phone numbers
Swift 5.3 framework for parsing, formatting and validating international phone numbers. Inspired by Google's libphonenumber. Import PhoneNumberKit at the top of the Swift file that will interact with a phone number. All of your interactions with PhoneNumberKit happen through a PhoneNumberKit object. The first step you should take is to allocate one. A PhoneNumberKit instance is relatively expensive to allocate (it parses the metadata and keeps it in memory for the object's lifecycle), you should try and make sure PhoneNumberKit is allocated once and deallocated when no longer needed. To parse a string, use the parse function. The region code is automatically computed but can be overridden if needed. PhoneNumberKit automatically does a hard type validation to ensure that the object created is valid, this can be quite costly performance-wise and can be turned off if needed. -
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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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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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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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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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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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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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SIM Card Manager
Windows tool to read SIM card information and content
SIMCardManager is Windows tool that can read a SIM card information (ICCID, IMSI, Pin states) and SIM card content : phone book (ADN), Fixed dialing (FDN) and SMS messages. It also enables you to authenticate with PIN when activated and export SIM card content to text files. You need to have a PC/SC compatible smart card reader or dongle with SIM slot. You can find other useful tools on my website http://www.idrix.fr -
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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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BlackBelt Privacy Tor/i2p+WASTE+VidVoIP
Browse on Tor/i2p, Anon p2p Chat / FileTx, Conf / Video VoIP
Open Source - GPLv3 inc images. *** PLEASE NOTE: There are now 2 seperate versions here. *** One is Pre Firefox 57. The other is Post Firefox 57. *** For those providing mirrors, please enable your users to realize this. Vidalia Based, Tor as a Service Solution. MicroSip: enables FREE PC to PC video calling with no account sign-up and no middleman server. WASTE: enables FREE Conference VoIP, chat, file transfer and support. *** AI Powered *** Tor/i2p: enables safer browsing. Tor/i2p Profile: Browse over Tor/i2p on Firefox (Vanilla, ESR), Waterfox (Classic, New), Palemoon , LibreWolf and legacy CyberFox. As with all versions of Tor - do not rely on this for strong anonymity. A usability enhanced Privacy Pack. Use alongside Tor Browser Bundle. An installer, for : Vista 32/64, Win7 32/64, Win8 32/64, Win10 32/64, Win11, Linux (Wine) -
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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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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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QKSMS
A beautiful SMS messenger for Android
Fall in love with texting all over again with QKSMS! QKSMS is a beautiful, simple and customizable open source replacement to the stock messaging app on Android. It's got a clean, clutter-free intuitive design that lets you focus on what matters, plus millions of colors to choose from to make the theme of the entire app truly your own. QKSMS has some powerful features, including photo sharing, group chat, conversation blocking, message backup and restore and more. It's got everything you need from a text messaging app, plus all the fun features that make for a great text messaging experience. Simply put, QKSMS makes texting magical again, just the way it should be! -
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SPICE Program Applications
The USAID-funded SPICE program, implemented by the Open Institute.
The USAID-funded SPICE program, implemented by the Cambodian NGO Open Institute, facilitated from 2012 to 2015 social and business innovation using communications technology and improved communications in Khmer via mobile devices. The purpose of the program was to build on partnerships between technology-oriented civil society organizations (CSOs) and private sector entities to utilize existing technologies to promote and deliver a greater diversity of information to the public, improve communications in Khmer using mobile devices, and increase the capacity of CSOs to use social media for development purposes. Other documents of the SPICE program can be found in http://spiceprogram.sourceforge.net -
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OpenSIPS (former OpenSER) is an GPL implementation of a multi-functionality SIP Server that targets to deliver a high-level technical solution (performance, security and quality) to be used in professional SIP server platforms. IMPORTANT: this is no longer the main hosting for the project. This was moved on GITHUB - https://github.com/OpenSIPS/opensips
