Showing 21 open source projects for "information system"

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  • 1

    oreka

    Enterprise telephony recording and retrieval system

    Enterprise telephony recording and retrieval system with web based user interface. The project currently supports recording voice from VoIP SIP, Cisco Skinny (aka SCCP), raw RTP and audio sound device and runs on multiple operating systems and database systems. It can record audio from most PBX and telephony systems such as BroadWorks, Metaswitch, Asterisk, FreeSwitch, OpenSIPS, Avaya, Nortel, Mitel, Siemens, Cisco Call Manager, Cosmocom, NEC, etc... It is amongst others being used in Call...
    Downloads: 20 This Week
    Last Update:
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  • 2
    MCU Media Server

    MCU Media Server

    SIP Video Multiconference Media Server with WebRTC support.

    REPOSITORY MOVED TO GITHUB!! https://github.com/medooze/media-server Video Multiconference Media Server with WebRTC support. Provide Multiconference and video broadcasting services to any SIP service. Supports VP8, H264, MP4V-ES, H263 and H263P, continuous presence, RTMP flash broadcasting, adhoc conferences, load balancing and administrative WEB interface. JSR309 driver implementation under development. .
    Downloads: 2 This Week
    Last Update:
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  • 3

    ss7mon

    ss7mon an SS7/SIGTRAN signalling monitor

    ss7mon will implement the following features for SIGTRAN and SS7 family of protocols: - Message sniffer. - Call flow tracing. - Traces DB. - Distributed tracing. - QoS monitoring. - KPI monitoring. - Web interface. - CLI command line interface. - ...etc.
    Downloads: 0 This Week
    Last Update:
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  • 4

    cdr2serial

    Send Asterisk CDR to serial device

    cdr2serial sends Asterisk CDR serial port for the use of software for hotels. They have been used libraries asterisk-java for agi events and jssc for send via serial port.
    Downloads: 0 This Week
    Last Update:
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  • 5
    Process flow designer that creates applications (Saflets) for us in the SafiServer engine. SafiWorkshop is used for the design, real time debugging and deployment of advanced flow based applications. Currently tailored for use with Asterisk PBX.
    Downloads: 0 This Week
    Last Update:
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  • 6
    A telephone interface to real-time arrival information for buses in the Metro King County transit system.
    Downloads: 0 This Week
    Last Update:
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  • 7
    The Genalyze log analyzer, a simple yet highly extensible framework for parsing log files, was designed for parsing T-Server log files generated by Genesys contact center software platform, yet is generic and easy enough to adapt for other log formats.
    Downloads: 1 This Week
    Last Update:
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  • 8
    Prototype testbed implementation of the IETF Media Server Control (MEDIACTRL) SIP Control Framework, comprehensive of both control and processing functionality (as in IMS MRF, Media Resource Function).
    Downloads: 0 This Week
    Last Update:
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  • 9
    SipManager is a full-feature IP-Centrex PBX web app for hosted/multi-client PBX based on Asterisk and OpenSer. VoIP/ToIP platform (PBX, Fax, workflows, etc.). User-friendly web interface. Realtime architecture with SQL database. ASP. http://www.ovvoe.com
    Downloads: 1 This Week
    Last Update:
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  • 10
    Sipana is a distributed SIP monitoring tool to monitor the SIP signaling behavior using sequence diagrams and providing SIP end-to-end performance metrics through a centralized Web interface. For more information please visit http://sipana.org/
    Downloads: 0 This Week
    Last Update:
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  • 11
    Jabast connects XMPP Clients to a asterisk PBX to allow extension listening, calls and presence controle for Agents.
    Downloads: 0 This Week
    Last Update:
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  • 12
    Compiere-NMA is a compiere module for network monitoring, pbx cdr, accounting and billing. Target is to provide the needed infrastructure for providers and companies needed to monitor and bill IP networks based on Compiere ERP + CRM.
    Downloads: 0 This Week
    Last Update:
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  • 13
    This is a Java wrapper for Ffmpeg. The JMF (Java Media Framework) plugins system lets one use JMStudio or other application to play H.263/rfc2190 streams. Support for H.263+ and MPEG-4 is currently under test, please contact us for further information.
    Downloads: 0 This Week
    Last Update:
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  • 14
    An more efficient alternative for writing complex Asterisk dial plan files, without using the cumbersome line numbering scheme.
    Downloads: 0 This Week
    Last Update:
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  • 15
    Time of day service over telephone using Voicent Gateway, a VoiceXML gateway that specially designed for voice modems. A Free version is available for download at http://www.voicent.com/download. Sample code for interactive telephony applications.
    Downloads: 0 This Week
    Last Update:
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  • 16
    The purpose of this project is to create an open source voicemail/unified messaging system that runs on Linux that has the ability to integrate with business telephone systems.
    Downloads: 0 This Week
    Last Update:
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  • 17
    Dreadnought is a client/server application to configure the Open Source Linux PBX Asterisk.
    Downloads: 0 This Week
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  • 18
    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
    Last Update:
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  • 19
    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
    Last Update:
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  • 20
    Web Callback Portlet is a portlet to be used with Liferay portal (other JSR 186 compliant portals might also work). Customers are entering information into a form and are put intouch, via phone, with a customer representative.
    Downloads: 0 This Week
    Last Update:
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  • 21
    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. Existing SIP solutions for the browser are forced to use the WebSockets API to...
    Downloads: 0 This Week
    Last Update:
    See Project
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