Search Results for "linux live 32"

Open Source - GPLv3 inc images. A WASTE client. Download and create your own WASTE networks. Move 1000's of GB's at 100MB+ per sec. (800 Mbits per sec) FULL pause and resume capable. Voice Conference, Chat, Transfer files and Participate in Forums in a secure environment. For Windows XP 32/64, Vista 32/64, Win7 32/64, Win8 32/64, Win 10, Win 11, Linux (WINE). *** User Based Access Control - for voice, chats, file transfers and uploads. (useful in NULLNETS) *** Distributed...

If you want to quickly set up a multi-tenant PBX on matter of few minutes, this is a suitable choice. - Demo environment. You can start to add extensions, gateways and dialplans, and quickly have a fully working PBX environment. - Import previous FusionPBX backup to Live environment. After this you will have a quickly working PBX on few minutes. - Install on hard disk. With this option you can install the environment to Hard Disk. Username and password for PostgreSQL is fusionpbx...

From Null to a Full Featured PBX in matter of minutes!! If you want: - To boot or install quickly Kazoo on your server. - To test about how it works, without installing on your hard drive. - To install on cluster environment - To install if your previous platform go downtime. This is a suitable option. What is included? Monster-UI Open Source Apps, like SmartPBX, CallFlows, PBX Connector, Voicemails, Faxes, Accounts and Number Manager Kazoo v4.3.x Kamailio ...

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. .

Source code content for Open Source packages used by the ARRIS SecureMedia Encryptonite Server, which provides a streamlined process for service providers to ingest, transcode and securely deliver live and on-demand content.

Project moved to GitHub. https://github.com/davidfischer-ch/smpte2022lib

Libre Software to manage audio/video connection from a PC to a libre hardware real time stethoscope hardware, developed to allow doctors make a remote diagnose in real time in isolated rural areas of developing countries or for medicine teaching

Zanzibar is a complete, standards based IVR. It includes an MRCPv2 Server with ASR and TTS engines as well as an voiceXML interpreter so that you can deploy and run voiceXML applications. It integrates with VOIP PBX’s (like Asterisk) using SIP and RTP.

Video Applications for Asterisk (libh43m, app_h324m, app_mp4, app_rtsp, app_transcode, app_swf) H324m stack, mp4 record and playblack, rtsp support, video transcoding and swf file playback.

Cairo sets out to provide an enterprise grade, MRCPv2 compliant speech solution utilizing existing open source speech resources such as FreeTTS and Sphinx-4.

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.

A Client-Server application to make votings with life evaluation via Bluetooth. Server is written in Java for use an any platform. Client is a Java-midlet for use on most mobile phones. It is result of a TU-Ilmenau software project. (See the Credits)

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.

JSR180 is the standard API spec by jcp.org which is been implemented over Resiprocate Stack. Our purpose is to enable "Open Source Code with Open APIs". By implementing JSR180 we could easily access the SIP Stack by the API documented in the JSR180 Spe

Development of software and alternative firmware for Asterisk-appliance D-Link HorstBox (DVA-G3342SB), free kubuntu live-dvd for firmware development, sourcecode (GPL) for original D-Link firmware, modifications/patches.

IHU is a VoIP application for Linux (using Qt and Speex), with low latency, crypted stream, minimal use of bandwith, and without intermediary servers. It is the easiest way to talk real-time with your friends (like phone) on the internet or LAN.

ST-PBX Live is a GNU/Debian Linux distribution integrating Asterisk, freePBX and other tools to try from CD or install in HD a complete VoIP system.

rtusd is a foundation set of real time enabled User Space Devices/Drivers designed to link to HRT high resolution Timer controlled POSIX threads and link re-factored Linux hdw drivers for 2.6.18+. rtusd propogates Linux RT to user space generically.

The MRCPv2 protocol is designed to allow client devices to control media processing resources, such as speech recognition engines. MRCP4J provides a Java API that encapsulates the MRCPv2 protocol and can be used to implement MRCP clients and/or servers.

Utilities enhance AMP, A@H, Asterisk and Cisco XML. Speed Dialing revised using web interface. Paging using custom meetme routines. Intercom by dialing 0 + Ext #. Browsing AMP's extensions on your phones LCD. Backup patch for Asterisk@Home maint screen.

Kasterx is a live-cd knoppix/debian based distro with asterisk preconfigured with AMP (asterisk management portal). Designed with new users to asterisk in mind to jumpstart you into the world of using linux as a pbx/voip system.

RUMPI: Real Time Multi Peer Interaction. RUMPI want to be as ubiquitous as Flash-browser-plugin, but enabling browsers for real time communication between peers: VoIP (IAX2) and Real-time video Streaming. RUMPI is not a plattform, is a Browser plugin,

Global mesh development software and hardware platform from developing global wireless mesh network using multiple frequencies.