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Kablink open team collaboration software uses social networking to unify team workspaces w/ real-time web conferencing. Collaboration for knowledge networking, program management, communities-of-practice, telework, other business process/functional areas

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.

The OSP Toolkit is a client side implementation of the ETSI OSP VoIP Peering protocol (ETSI TS 101 321). The OSP Toolkit project was begun in 1998 and the code has been incorporated into many commercial and open source VoIP products.

VoiceOne gives you the ability to install and compleatly configure a pbx platform based on Asterisk 1.8 with an easy to use web GUI, which would be a framework to build a communication server adding various plugins.

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

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).

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.

AGIS stands for Asterisk Groupware Integration Server. It provides a SOAP backend for access and configuration of the Asterisk PBX. Further it implements a SOAP based Asterisk application for E-Groupware with management functions for users and admins.

An open source software project delivering converged voice and data applications. The StarPound platform enables you to rapidly develop voice and data solutions using the power of business process modeling. More info: http://www.starpound.org.

Session Initialtion Protocol (SIP) is widely used for telephone services over the Internet. CafeSip provides a suite of open-source tools and applications for creating customized SIP services and applications using the Java.

tcliaxc is a Tcl extension which provides you access to iax voip protocol via iaxclient library. tcliaxc lets you use a Tcl package to do basic iax operations such as registering to a server, make a call, handle iax event and so on.

SIP UMS is an Unified Messaging Server for the SEMS/IVR. It is an extendable voicemai and IVR platform written in Perl. It includes a web based adminstration tool, voice prompts (in english) and sample scripts that will work with SEMS/IVR.

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.

Open UC Framework (openucf) is a unified communication server which supports telephony, presence, instant messaging and contact management.

The minimum hardware to have a portable IP-Phone. A VoIP telephone that is simple to assemble and an emulator for portability. Core runs under Linux. Modular functionalities: a display, keys (dialpad, DND,...), a web server. Free, simple and useful.

The Asterisk .NET library consists of a set of C# classes that allow you to easily build applications that interact with an Asterisk PBX Server (1.0/1.2/1.4 version). Both FastAGI and Manager API supported. .NET/Mono compatible.

MumurWeb will be a comprehensive webinterface for Murmur servers based on PHP. Murmur is the server component for the Open-Source voice communication Mumble.

Prototype testbed implementation of the IETF Media Server Control (MEDIACTRL) SIP Control Framework, comprehensive of both control and processing functionality (as in IMS MRF, Media Resource Function).

talkLock is voice encryption software for your cell phone. talkLock includes a server component so that you own all points of encryption and decryption of your communications.

CHAROZT is an Asterisk MOH (Music on Hold) Uploader. It is a Linux-based application installed in Linux desktops that converts MP3 to WAV for Asterisk supported MOH. It can also upload converted files to the Asterisk Server.

AMS is a suite of software intended to make day to day administration and monitoring of an Asterisk PBX server easier. It contains a daemon that acts as a proxy to Asterisk's Manger Interface and a GTK GUI application for monitoring and administration.

Le stelib sono librerie php per costruire dinamicamente pagine web per PC, palmari, cellulari o telefoni IP; queste librerie sono un insieme di classi che ci aiutano a gestire database, PABX VoIP (Asterisk), mootools , phpPlot etc.

trakkcor tracks your position via gps and submits the position data to your server. All you need is a j2me-compliant mobilephone which supports JSR-82 and an external gps receiver.

VoIP Toolkit / Call Control with Integrated Media. High-level Java API for creating SIP enabled VoIP applications. Suitable for either the desktop (softphone, phone applet, incoming call gatekeeper) or server-side (auto attendant, ACD, voicemail).