Search Results for "gnu/linux" - Page 6
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Showing 257 open source projects for "gnu/linux"
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VOIP client/server in python >= 2.6. Audio in/out: ossaudiodev (UNIX like) or SoX. Network: bzip2 compression, speex or ogg audio compression, you can configure all, minimum bytes per second: 350-400 in speex mode: U8, 6 kHz, quality 0, bzip2, buf 4K
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This is a open source media gateway controller. The protocol used is MEGACO/H.248. In a later version also SIP will be supported.
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FreePBX module for ENUM lookup API at enumplus.org.
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Universal Queue and Unified Communications core implementation upon the Platform as a Service paradigm. An abstraction layer for the hardware communications platform among a plugin based model brings independence from technology and media to the project
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A $10 Analog Telephone Adapter (ATA) that allows low cost routers such as the WRT54G to connect to an analog telephone. The application is low cost VOIP, in particular for the developing world.
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FreeSentral is a full IP PBX consisting of a Linux Distribution, an IP PBX and a Web Graphical User Interface for easy configuration.
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This project is aim to develop module named app_avi.so for asterisk to add the ability of play avi file.
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app_swift provides an interface into the Cepstral TTS libraries to stream text-to-speech audio from the Asterisk PBX without creating files.
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Open UC Framework (openucf) is a unified communication server which supports telephony, presence, instant messaging and contact management.
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SIP Inspector is a tool written in JAVA to simulate different SIP messages and scenarios. You can create your own SIP signaling scenarios, customize SIP messages and monitor incoming and outgoing messages. The tool can play RTP streams from a pcap fi
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The goal of the project is to create a high-performance, open-source and standards-compliant implementation of a Home-Subscriber-Server (HSS) for use in a IMS context.
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SEMS is a GPLv2+ media and application server for SIP based VoIP services. Basic applications like announcement, pre-call announcement, RBT, conference, voicemail, mailbox etc. and lots of example applications available.
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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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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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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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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.
