Search Results for "gnu/linux" - Page 23
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Showing 1356 open source projects for "gnu/linux"
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SkypeXMPP is a Karaka™ fork aimed to provide Skype to XMPP gateway It supports presence, instant messaging, multi-user chat and voice (thanks to Freeswitch) This product uses the Skype™ API but is not endorsed or certified by Skype.
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Twisted protocols providing access to the Asterisk PBX's AMI and FastAGI interfaces from within the same programming process.
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DTMF detector library and/or application that reads in the specified audio file and returns/outputs whether DTMF detected and/or list of detected digits. May have GUI and command line interfaces.
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The start of an alternative to Asterisk.NET. The goal of this project is to create a manager connection library that simplifies interfacing with multiple Asterisk servers. Asterisk Connection Manager is currently very new and incomplete and undocumented.
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PHPAGI is a PHP Class for writing AGI applications for use with the open source Asterisk PBX software.
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Use this when attachments aren't offered with your email service. (i.e. Proxy Mail). Simply Supply a link to download the 11KB Audio Player and Copy/Paste your Message into the Email. See Website For Demo. http://audiotext.sourceforge.net
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OpenBTS
GSM L1-L3 stack with SIP network interfaces
OpenBTS is an implementation of the GSM air interface (Um) that allows cellular handsets to be used directly as SIP endpoints. It uses a software-defined radio to generate its air interface and uses Asterisk or yate as its network interface. -
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[es ] Aplicación web central para particulares. [en ] Core web application for individuals. [fr] Application Web centrale pour les personnes. [pt] Central de aplicação web para indivíduos. [ar] المركزي تطبيق ويب للأفراد
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THUNDER!Box synchronisiert das Thunderbird-Adressbuch an einen FRITZ!Box DSL-Modem/Router. --- THUNDER!Box synchronizes the Thunderbird addressbook to the FRITZ!Box DSL-Modem/Router.
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Java GUI for editing the Fritz!Box phone book.
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Simu server project is aimed for IP telephony (VoIP) between classical SIP phones and Flash streaming applications.
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Fluent is a suite of tools for the management of diverse VoIP networks. Fluent RTE, a reporting and templating engine, is at the core. Add-on modules are available to manage provisioning, CDR, feature sets, routing and more.
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Open source software components of the AastraLink Pro 160 Supplied from CDROM "71-001049-02 REV01" by Aastra Telecom
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An open source CTI client for Asterisk using the Asterisk Manager API. Uses pygtk and pyAsterisk.
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nohandcom is an accessibility software running on mobile devices like smartphones and PDAs that allows physically handicapped users complete access to all device functionality by emulating either key or touchscreen input.
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SIP.NET is SIP stack .NET library written in C#. SIP message parser realized on Deterministic Finite-state Machine (DFA parse all SIP message for one pass). DFA generates automatically from Augmented Backus–Naur Form from RFC3261.
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This is an initiative to assemble/develop the cheapest, easiest to setup, easiest to manage, scalable, Open Source, standards-based, wireless local do-it-yourself telephone company toolkit in the world.
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It is an appliance software that integrates the best tools available for Asterisk-based PBXs into a single, easy-to-use interface. include astercrm astercc voicemail,fax,et
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Live-cd com Servidor SIP baseado em opensips. O Servidor de Telefone Livre transformará um simples PC numa central SIP multidominio. Estamos construindo a documentacao do projeto na wiki do sistema trac.
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This is mobile SNS application. It will build a bridge between you and your friend. You can share everything what you have interesting. Share your activity, you friend can response it.. ShareLife run on 2.0+ platform.
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Castadiva is a test bed based on low-cost off-the-shelf devices, which is used to test protocols developed for MANETs. Castadiva is completely compatible with the file format used by the ns-2 simulator.
