Search Results for "front-end"

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

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

eGW-pbx is a eGroupware Module to configure and use Asterisk PBX.

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.

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.

Activa brings the Asterisk IP PBX to the call center. Built on top of Asterisk, Activa components enable successful call center implementations adding value in areas such as computer telephony, screenpop&click2dial, agent control, automatic dialing...

Intel Integrated Performace Primitives audio/video codecs plug-in for the OPAL/OpenH323 library including G.728, G.729, G.723.1, G.722.2 GSM-FR, GSM-AMR, H.261, H.263, H.264 and MPEG4 part 2.

The project has moved! Please find current versions at https://www.gnugk.org/ The GNU Gatekeeper (GnuGk) is a full featured H.323 gatekeeper under GPL license. It supports VoIP and videoconferencing and includes Radius and database support and many call routing functions. The project has moved! Please find current versions at https://www.gnugk.org/

PLEASE NOTE that all current AstLinux development is on GitHub: https://github.com/astlinux-project/astlinux Thanks to SourceForge for providing free hosting to the AstLinux Project for more than 10 years! Old source archive remains on SourceForge. AstLinux is a custom Linux distro centered around Asterisk, the Open Source PBX. AstLinux has many features that make it ideal for embedded and commercial Asterisk based solutions.

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.

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 .

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

M2S is an application created by Jaume Segarra web-based, uses HTML5 code, PHP, Javascript and CSS to provide users of Firefox OS an application to send free messages. Being a web application, M2S is available for all devices with Internet access either mobile devices, computers or televisions SmarTV.

jlibrtp aims to create a library that makes it easy to support RTP (RFC 3550,3551) in Java applications. SRTP (RFC 3771) has been delayed in favor of RFC 4585.

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.

skiputler makes your Topcom "Butler 4012 USB VoIP" phone work on Linux and Mac OS X. It's a userspace driver which connects the phone to Skype.

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.

One of the first secure telephony programs. Not limited to VOIP but can also directly encrypt a telephone line by using modems. Calls are encrypted using Diffie-Hellmann key exchange and several symmetric ciphers. Download is possible from Germany only, see homepage

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).

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.

Voice over IP (VoIP) software specifically for ham radio operators. The primary purpose is to use the Internet to link Ham repeaters with a secondary purpose of providing remote access for users. Compatibility with EchoLink and IRLP is a goal.

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.

asterCRM is a call center software for asterisk based VoIP system, also it has some CRM functions. It provide useful features such as pop-up, predictive dialer, click to call, extension status .... astercrm could work with all asterisk based system.