Search Results for "routing"
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Showing 8 open source projects for "routing"
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GNU Gatekeeper (GnuGk)
H.323 Gatekeeper for VoIP and videconferencing
...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/ -
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PBXlab adds value to business thanks to its valuable contribution of technological solutions. PBXlab promotes the use of free software programs that can be tailored to the real needs of its users, speed of movement between users, at a very low cost or no cost. PBXlab offers development of add-ons and improvements to programs, leading to more full, safe and adapted to customer needs programs. We offer technological independence without having to develop products from...
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VozBox. Más que una centralita.
VozBox. Asterisk VoIP PBX.
VozBox es poder disponer de un Sistema de Comunicación Avanzado. Es poder llegar más lejos. Es poder convertir lo más sofisticado en lo más sencillo. Es poder contar con la más avanzada tecnología de comunicación, reduciendo sus gastos. Es una centralita VoIP Asterisk con altas prestaciones para las Empresas. Última versión disponible 2.11.19 en la web de VozBox: http://www.vozbox.es/descarga/ Internet connection required during installation. Update version: 2.11.19 1 -... -
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Git repo: https://github.com/asipto/siremis Web management interface for Kamailio (OpenSER) - handle subscriber profiles, access control lists, accounting, least cost routing and load balancing, monitoring charts, xmlrpc communication with SIP server
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Fluent is a suite of tools for the management of diverse VoIP networks. Fluent RTE, a reporting and templating engine, is at the core. Add-on modules are available to manage provisioning, CDR, feature sets, routing and more.
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jSipgateConnect library allows Java-based access to all functions specified in the 'Sip Advanced Messaging, Usability, Routing and Accounting Interface' specification.
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astrtr
Asterisk module to route traffic (very obsolete)
astrtr is a FreePBX module for Asterisk to route calls from one trunk to another. Makes Asterisk PBX a VoIP Switch as well. Includes multiple routing tables, max connections per trunks, some reports, CDRs, etc. -
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sippet
C++ SIP stack based on Chrome source code
This C++ library has been designed as a Chrome SIP stack. Sippet is an open-source SIP User-Agent library, compliant with the IETF RFC 3261 specification. It can be used as a building block for SIP client software for uses such as VoIP, IM, and many other real-time and P2P communication services. The main target was to enable Javascript applications to use UDP, TCP and TLS transports along WebSocket. Existing SIP solutions for the browser are forced to use the WebSockets API to...
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