Open Source ChromeOS Internet Phone Software

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.

Mumble is an open source, low-latency, high quality voice chat software primarily intended for use while gaming. It includes game linking, so voice from other players comes from the direction of their characters, and has echo cancellation so the sound from your loudspeakers won't be audible to other players.

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

Peers is a very simple softphone. It's a SIP User-Agent, written in java, it works on windows, linux and mac. It can be used with SIP servers like opensips or asterisk IPBX. It supports G711 codec (PCMU and PCMA) and telephone-events (DTMF).

wxCommunicator is a cross platform SIP softphone written in C++ utilizing customized sipXtapi user agent library and wxWidgets 2.8.9 GUI library. For a list of supported features see http://wxcommunicator.sourceforge.net/features.html .

With SIP Proxy you will have the opportunity to eavesdrop and manipulate SIP traffic. Furthermore, predefined security test cases can be executed to find weak spots in VoIP devices. Security analysts can add and execute custom test cases.

GibPhone is a highly extensible VoIP/IM client for the .NET framework that uses a powerful plugin engine to allow for UI extensions and any call control stack / media payload / transport protocol combination.

Simple, small, secure web conferencing on any platform that supports Java.

Utilities enhance AMP, A@H, Asterisk and Cisco XML. Speed Dialing revised using web interface. Paging using custom meetme routines. Intercom by dialing 0 + Ext #. Browsing AMP's extensions on your phones LCD. Backup patch for Asterisk@Home maint screen.

Voxy - the simplest way to integrate VoiceXML applications in Asterisk. Configure your dial plan with the URL of your VoiceXML application and it's done.

This project contains a SIP stack and server applications built on top of the stack, examples of which are: Stateless Proxy, Registrar,NAT Keepalives,MWI Notifier server. The server applications have been designed to work in association with Asterisk.

OpenGatekeeper is an Open Source H.323 Gatekeeper based on the work done by the OpenH323 project.

A Sip live audio feeding agent. The agent captures audio from sound card and sends live audio stream(uLaw) to caller(sip phone) using RTP. It is based on Peers 0.3(http://peers.sourceforge.net/). Can be used in IP telephony to broadcast live audio.

dtmfbox is a small softswitch application (SIP/CAPI), that can be used to control different tasks over telephone keyboard via DTMF. Mostly, it was made to run on the AVM FRITZ!Box 7170 (mipsel) but works under Unix/Linux and Windows, too.

Maintained version of old asterisk applications ported to 1.4 that for copyright reasons cannot be included into official Digium's releases and that for some reason the author is not keeping up-to-date. The project also require Zaptel instead of DAHDI.

Open Source Voip Billing / Calling Cards / Callshops / LCR For Asterisk

Teamspeak enables people to speak with one another over the Internet. The Administration of a Server can be done by Telnet, Mysql, Web-Frontend or with the Teamspeak-Client. This Project is a Interface to Perl.

Cross-platform Java Teamspeak client

Real-Time Collaboration (RTC) and conferencing project. This product has been deprecated. Please see the replacement product, CoolCollaborator, at http://coolcollaboratr.sourceforge.net/

Asterisk Admin is a php/ajax web app to manage realtime asterisk deployments. It supports both static and realtime configuration, and can manage any and all features found in asterisk 1.2+ that can be controlled through asterisk's realtime interface.

This simple php page show a list of queues and his members and agentes activity by quering Astrerisk through the Manager Interface, reloading itself every n seconds. Suggestions are welcome.

ISDN CAPI Channel driver (chan_capi) for the Asterisk Open Source VoIP Platform. THIS PROJECT HAS BEEN MOVED TO chan-capi.org

A C++ framework utilizing Design Patterns for creating Linux and Windows communications applications that contain Dialogic® products. Includes media and network classes (analog, digital, SIP, H323), multithreaded event handling, distributed app support.

Asterisk Manage API and FAGI C++ Framework. Implemented with high efficient OS independent Sockets wrapper (ACE). Provide C++ API wrappers for Asterisk Manager API and FAGI. A Manage API relay(proxy) server and a FAGI socket server will be implemented.

The Cisco IP Phone Idle Display Logo Engine project or CIP IDLE A framework for displaying custom logos on Cisco IP Phones 79XX series. Also provides a Java alternative to cip2gif.exe and gif2cip.exe