Projects tagged ‘conferencing’ and ‘voip’

Asterisk is a complete PBX and telephony toolkit in software. It runs on Linux, *BSD, MacOSX, and Solaris. It provides all of the features you would expect from a PBX and more as it enables ... [More] developers to build customized voice applications of many types. Asterisk does voice over IP in many protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. [Less]

Ekiga (formely known as GnomeMeeting) is an open source VoIP and video conferencing application for GNOME. Ekiga uses both the H.323 and SIP protocols. It supports many audio and video codecs, and is ... [More] interoperable with other SIP compliant software and also with Microsoft NetMeeting. [Less]

Yet Another Telephony Engine is a next-generation telephony engine; while currently focused on Voice over Internet Protocol (VoIP) and PSTN, its power lies in its ability to be easily extended. Voice ... [More] , video, data and instant messaging can all be unified under Yate's flexible routing engine, maximizing communications efficiency and minimizing infrastructure costs for businesses. [Less]

Sems is a extensible media server which helps you adding voice services to your VoIP system.

Gemeinschaft is the open-source PBX based on Asterisk, MySQL, Apache, and PHP, and is designed for high availability and clustering. It provides automatic provisioning for mass deployment, and can ... [More] handle over 10,000 users. Administration is done via shell scripts or a Web GUI. Hot-desking and mobility are supported. German voice prompts are included. There is a standards-compliant Web GUI with internationalization (and German and English translations). Outbound and inbound routing with full PCRE support is included. [Less]

The GNU Gatekeeper (GnuGk) is a full featured H.323 gatekeeper under GPL license. It includes Radius and database support and many VOIP routing functions.

Gemeinschaft (by Amooma) is the open-source PBX based on Asterisk, MySQL, Apache, and PHP, and is designed for high availability and clustering. It provides automatic provisioning for mass deployment ... [More] , and can handle over 10,000 users. Administration is done via shell scripts or a Web GUI. Hot-desking and mobility are supported. German voice prompts are included. There is a standards-compliant Web GUI with internationalization (and German and English translations). Outbound and inbound routing with full PCRE support is included. [Less]

SipStream is an open source soft-phone being designed at Carleton University. It is or hope that this phone will be to provide an open source alternative to VoIP as a web application. The primary ... [More] purpose of this project is the development of integrated VoIP capabilities of the BigBlueButton web conferencing application. Consequently, we hope to further develop a standalone application of this phone capable of being run on any web service. To do so we will be designing this application to run on a red5 using open source tools. If you would like to receive a copy of our project report, feel free to contact one of the group members. These are previews of the phone module implemented into BigBlueButton. This is a preview of the standalone phone interface that we have implemented using Flex. [Less]

Gateway between sip and http. Uses RESTful APIs on http side.

Web-based graphical user interface (GUI) for FreeSWITCH telephony switching platform. Provides configuration tools for standard functions such as configuring phones, conference devices, ATAs ... [More] , dialplans, SIP providers, and so on. MVC framework is utilized [Less]