Search Results for "source code for preload" - Page 4
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Showing 83 open source projects for "source code for preload"
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A fully perl written sofware for windows (also stand alone binary code included, so it does not require perl installation to use it) to receive short messages (SMS) sent by mobilephone.
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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.
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BeepagePHP provides a PHP-based web client to the beepage client/server paging system. Beepage is a Unix-based, Internet-aware text paging system that implements the Telocator Alphanumeric Protocol (TAP). Source code is hosted at http://git.io/rgikgg
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The purpose of this project is creating a plugin for <a href="http://www.ethereal.com/">Ethereal</a> that will enable it to decode ITU-T Recommendation H.323 traffic. The implementation is based on <a href="http://www.openh323.org">OpenH323</a> code
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An Open H.323 based channel driver for the the Asterisk PBX. This code has now been intergrated into the Asterisk source tree. Please see the Asterisk source tree for any future updates of this software.
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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
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Chopped LIVR
Generate Asterisk IVR dialplans
Chopped LIVR takes YAML input and generates Asterisk dialplans. It is primarily useful for generating IVR (Interactive Voice Response) applications; however, it may be used as a tool to generate any Asterisk dialplan. Chooped LIVR also incorporates control structures and functionality enhancements to the base Asterisk dialplan language. -
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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...
