PNX system develops the middle-ware source code to glue Asterisk with a number of powerful telephony products such as: 1) OpenH323 H.323 stack 2) Vovida SIP stack 3) Bayonne Voice Automation Platform The advantages? An advanced IP PBX supporting the wide
Siphon will ultimately be a Software Voice-Over-IP phone using the SIP protocol. It will support quicknet cards as well as traditionnal soundcards. It is developped for Linux (gtk) and the win32 platform.
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SipSpy is a distributed monitoring tool for SIP networks. SpyAgents run on each of the nodes to be monitored, and a SipSpy connects to each of these nodes, receiving information and displaying it in real-time for all the SIP packets monitored.
The G.O.N.E. is a softphone (or soft phone) running over the web, fully multi-plataform, it implements the SIP protocol, and is built to work on any SIP server, like Asterisk, and others. GONE will work on a complex sistem, but this will be showed a bit
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 send...
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SpITAssassin will be an enhancement to the successful SpamAssassin to detect Spam over Internet Telephony. This projects extends the SpamAssassin-functionality to insert SIP-compatible headers marking possible Spit-calls.
Current vTiger CRM integration is to control the Asterisk (soft) PBX to establish and log calls. This call is established by the vTiger user of the vTiger web based GUI. FreeSWITCH can be configured as a similar soft PBX and so the goal is to get vTiger to cause Freeswitch to establish SIP connections from the vTiger user's phone to the customer phone.
The SIP2IMS gateway aims to enable simple SIP clients to access IMS networks and provide support for IMS features such as Service-Routes, AKA authentication, IP-Sec/TLS etc.
Open Source SIP Back to Back User Agent) and Billing system. Has an Embeded TCL interpreter making session control and session mediation limitless in flexibility. Useful for: Routing Engine, Calling Cards, SBC, Peering, Wholesale and more!
uPBX aim is to give the user a GUI to manage all the interesting features available on the CallWeaver PBX software, such as T38, SIP, H323, PSTN, Conferences, queues and many more.
Tools and libraries for dialing URI addresses directly from telephone's numeric keypad w/o phone directories.
Encoded uri (for instance, SIP VoIP phone number) consists of decimal numbers only and can be entered from any telephone or other device.