Search Results for "using" - Page 3
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Showing 149 open source projects for "using"
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SafiServer is an engine that powers the applications (called Saflets) created using SafiWorkshop. SafiServer uses these applications to control communications (i.e. Asterisk) along with a multitude of other applications (database, email, etc).
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Rabboni VOIP is an Open Source Cross Platform Softphone developed with trolltech QT LIBRARY it uses PJSIP - Open Source SIP Stack
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This project is developed as a database application using Qt 4.0 and MySQL 5.0. Main aim of the project is store call details of each customer. At any time (usually end of month) admin can send call details to the selected customer.
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Outgrown Windows Task Scheduler?Windows Task Scheduler wasn't built for complex, cross-platform automation. Get a free diagnostic that shows exactly where things are failing and provides remediation recommendations. Interactive HTML report delivered in minutes.
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Quintum Tenor VoIP Gateway Call Detail Record (CDR) Advanced Server. This CDR Server can collect Call Detail Records (CDRs) from multiple Tenors. The data is collected simultaneously and continuously. Supported MySQL and file storages.
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OS SIP applet can be used to make a VoIP call from the web page. Integration can be made using java script API, which is provided by the SIP applet.
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Control your computer using a bluetooth enabled mobile phone. Move the mouse cursor send key strokes and control the most common applications such windows media player, internet explorer, firefox
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This is a SIP signaling layer to create a fully operative multipoint (video) conference server using SIP clients and RTP media streams in combination with strManager as a media management layer.
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MurMan, the Murmur Manager, is a user friendly, robust, and attractive web management interface for Murmur servers based on PHP, using Murmur's (server component for the Open-Source voice communication Mumble) Ice RCP.
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Run applications fast and securely in a fully managed environmentRun frontend and backend services, batch jobs, deploy websites and applications, and queue processing workloads without the need to manage infrastructure.
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Experimental open source project, which aims to test possibility of VoIP calls "from browser to browser". Main part is Java application implementing UDP hole punching, which is sent to both clients using JWS. See it online in action at project homepage
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Recqual is collection of scripts and patches using Asterisk and other Linux utilities to measure call quality on an automated basis.
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Add-on module to help with the deployment of SafiServer applications in Asterisk/FreePBX based systems. The module will allow administrators to reference Saflets (graphical call flow/IVR applications) using the FreePBX web interface.
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This project provides a meter of QoS on VoIP using the protocol SIP Media Loopback
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Twinkle is a VOIP based telephony software on linux. This project will install this software on MAC OS X using fink package management system. This will be included in stable tree of fink.
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MobileStratego is the online version of Stratego for mobile phones. It's developed with J2ME and it is a two players version. Online playing is achieved using peer-to-peer with virtual server or direct connection. Single phone is supported too
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A Sip live audio feeding agent. The agent captures audio from sound card and sends live audio stream(uLaw) to caller(sip phone) using RTP. It is based on Peers 0.3(http://peers.sourceforge.net/). Can be used in IP telephony to broadcast live audio.
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Avaya AES Bulk Administration tool. Allows adding and deleting of extensions in AES either by range (i.e. 4300-4352) or by using an comma-delimited file as a source.
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Configure your PC to access the Internet using the mobile and keep your connection charge under control. If program do not run install gambas2-gb-gtk gambas2-gb-qt packages. Additional language support for: Italian, French, Spanish, Croatian.
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Using the new C# Dll from swig portability of Openobex/ObexFtp to write a nice GUI able to transfer file between two devices with obex protocol. More Info’s here: http://cs-obexftp.wiki.sourceforge.net/
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PHP based Web-App amied at managing call-centers Outbound campaigns. Customizable call results and phone interview using external XML file.
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CDRisk is a project that aims to create a new CDR system, using AGI (Asterisk Gateway Interface) development, that works with transfers.
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J8051 is an API for communication between a 8051 microcontroller and a cell phone using Java ME, Bluetooth and "C" language callback functions, providing a easy way for embed systems developers build a communication between.
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Pops up a URL when a Call on an Asterisk Server is Answered, passing the Extension, Caller ID name and number. Connects to the Asterisk Server using the Manager Interface. Can be used to connect to a web based CRM.
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SIP to Skype Gateway/Bridge/Converter/Adapter. Make and receive SIP to Skype Calls and Skype to SIP Calls. Call Skype users using speed dial or use SkypeOut. Make SIP calls from Skype using a SIP provider or SIP PBX. Use as a Skype Trunk with a PBX.
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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/
