Search Results for "linux live 32"

...: Browse over Tor/i2p on Firefox (Vanilla, ESR), Waterfox (Classic, New), Palemoon , LibreWolf and legacy CyberFox. As with all versions of Tor - do not rely on this for strong anonymity. A usability enhanced Privacy Pack. An installer, for : Vista 32/64, Win7 32/64, Win8 32/64, Win10 32/64, Win11, Linux (Wine)

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 Autonomic...

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 v5.1 FreeSWITCH...

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 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.