Open Source Telephony Software

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

Connect your phone via WiFi, cable or bluetooth and you'll be surprised how easy and efficient it will be to manage your Android based smartphone with our software. Since it's first release MyPhoneExplorer evolved into the most popular freeware tool for smartphones. The software is constantly updated with new features. In opposite to many other solutions MyPhoneExplorer does work complete locally without using any third-party server!

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.

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)

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.

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.

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.

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

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.

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

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.

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.

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.

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

Remote control software for applications using wi-fi or bluetooth.

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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.

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. An installer, for : Vista 32/64, Win7 32/64, Win8 32/64, Win10 32/64, Win11, Linux (Wine)

pcapsipdump is libpcap-based SIP sniffer with per-call sorting capabilities. It writes SIP/RTP sessions to disk in a same format, as "tcpdump -w", but one file per SIP session (even if there is thousands of concurrent SIP sessions). Getting started: http://pcapsipdump.sf.net/

An Open Source (predictive) Dialer. OSDial is a full featured GPL dialer which does Manual, Power and Predictive dialing out of the box. Its easily installed through one of our .iso images. Install manual available at http://store.osdial.com/ Full details available at web site (http://osdial.com).

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

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 $

Vem ai o Novo Disc-OS Aguardem.. Uma Empresa Orkestral Estamos Voltando! Modificações: Asterisk 21 Nova Interface UI Centos 8 Nodejs Disc-OS é uma distribuição de um PABX IP baseado em software livre. Desenvolvido para o mercado brasileiro com interfaces em português, de fácil instalação e configuração, contendo Linux customizado, software Asterisk 1.4 e configurador Disc.

OpenLTE is an open source implementation of the 3GPP LTE specifications. Currently, octave code is available for test and simulation of downlink transmit and receive functionality and uplink PRACH transmit and receive functionality. In addition, GNU Radio applications are available for downlink transmit and receive to and from a file, downlink receive using rtl-sdr, HackRF, or USRP B2X0, LTE I/Q file recording using rtl-sdr, HackRF, or USRP B2X0, and a simple eNodeB using USRP B2X0. The current focus is on extending the capabilities of the GNU Radio applications and adding capabilities to the simple base station application (LTE_fdd_enodeb).