Showing 68 open source projects for "c source code"

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  • 1
    A set of JNI classes providing direct access to Asterisk PBX functionality from Java. This is not a socket-level interface into the Asterisk manager app but a true Java<->Asterisk integration at the Thread level.
    Downloads: 1 This Week
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  • 2
    Find out who's calling you before pick up the phone. No extra hardware is required as long as your modem is supported.
    Downloads: 0 This Week
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  • 3
    (A WebWorm Crashed my website!! :( )BlueWalk is a small software for Symbian, it is the first automatic system for personal communication, that allows to communicate with the people around you. It will give you a new way of using phone.
    Downloads: 0 This Week
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  • 4
    FreePhone is an easy to use IP phone application. FreePhone supports major platforms and protocols. Features include Conference calling, Voice mail and Chat rooms. A SDK is available for you to make additional features.
    Downloads: 0 This Week
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  • 5
    It's new technology for Development Personal Creation/Customization Services , based on XML standard and DAFSM (Finite State Machine Theory) , using multiple call control protocols.
    Downloads: 0 This Week
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  • 6
    OreIVR is a package of Java classes and command-line utilities that are used to access and control Dialogic computer-telephony hardware.
    Downloads: 0 This Week
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  • 7
    java telephony API, Dialogic's hardware support, telephony application IDE, mutlichannel telephony applications Server
    Downloads: 0 This Week
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  • 8
    The XTapi project is aimed at someday growing up to be a reference implementation for JTAPI. At this time it is not the intention to cover every JTAPI package, just the Core and Media packages are covered.
    Downloads: 0 This Week
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  • 9
    The CSTA Switch Simulator is a software telephone environment that can be used to test Computer Telephony Integration (CTI) applications. The simulator complies with the CSTA phase II specification.
    Downloads: 0 This Week
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  • 10
    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
    Downloads: 0 This Week
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  • 11
    We develop the PCPartner suite of ActiveX controls for integration of applications with Avaya Alchemy telephone systems over a network.
    Downloads: 0 This Week
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  • 12
    A system for logging/presenting information about incoming phonecalls. Currently only working for ISDN routers with the ability to send syslog messages.
    Downloads: 0 This Week
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  • 13
    SMSID provides a Linux PAM that allows the use of a GSM cellular phone with SMS capabilities as a hardware token in a strong authentication system. In addition to the Linux PAM module, an extensible SMS server is included. This server is written in Java
    Downloads: 0 This Week
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  • 14
    sippet

    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.
    Downloads: 0 This Week
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  • 15
    Open-ESME is an open source SMS client, intended to handle SMPP, EMI/UCP and CIMD traffic through the same interface.
    Downloads: 0 This Week
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  • 16
    OpenSVMS is an open source Short Voice Messaging System. SVMS (also known as VSMS or VoiceSMS) is a telecom service which enables users to send voice clips as messages rather than plain text messages.
    Downloads: 0 This Week
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  • 17
    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.
    Downloads: 0 This Week
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  • 18
    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
    Downloads: 0 This Week
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