Search Results for "debian-based linux" - Page 4

PJSGUA is an Open Source SIP UAC based on PJSUA reference UAC from PJSIP project

The project is a preconfigured VoIP PBX VM Image based on Asterisk. But provides more advance features, such as LDAP integration, VQ monitoring via RTCP XR reports, LDAP user login, integrated billing system and telephone directory PDF generator.

astrtr is a FreePBX module for Asterisk to route calls from one trunk to another. Makes Asterisk PBX a VoIP Switch as well. Includes multiple routing tables, max connections per trunks, some reports, CDRs, etc.

PHP based Viewer for Voice Servers like Mumble.

OaxRom WebSMS is a software that allow send multiple SMS from a web page, you will only need capture category, link with contacts and then you can send sms messages from a webpage selecting a category then will send sms to all contacts of the categor

I'm sad to say that, unfortunately, the serMyAdmin project is being discontinued starting in September 1st, 2010. After three years of development, the project no longer makes sense. The main reason to create the serMyAdmin was the lack of a GUI for OpenSER/OpenSIPS. The OpenSIPS control panel fulfill this need and it is not wise to maintain two GUI projects. Our team will support the opensips-cp initiative to avoid splitting the community between two open source projects for OpenSIPS GUI. I...

Open Unified Recording (OUR) is a full featured Linux based Open Source VoIP/SIP Call Recording engine, indexing and retrieval system. The system resides on the network and passively captures SIP sessions. Project is sponsored by UnifiedRecording.com

MurMan, the Murmur Manager, is a user friendly, robust, and attractive web management interface for Murmur servers based on PHP, using Murmur's (server component for the Open-Source voice communication Mumble) Ice RCP.

A web-based system that allows you to create groups of exten and generate reports of telephony usage, simple and easy to implement.

A modularized Linux platform optimized for embedded hardware hosting various network appliance packages (i.e. Router,Voip PBX,NAS,etc..) with an easy to use web interface for customization.

The system is used to control elements of a smart home remotely, with a web or a telephony interface. Automation can also be created to automatically execute actions based on events.

Fast & Lightweight PHP Based GUI For Asterisk opensource IP-PBX and doesnot requires any database. Designed to be used with Embedded Appliance or PC Based Solution. Its is different than Asterisk GUI & FreePBX ,no AJAX or MYSQL.

An XCAP server is used by XCAP clients to store data like buddy lists and presence policy in combination with a SIP Presence server that supports PUBLISH, SUBSCRIBE and NOTIFY methods to provide a complete SIP SIMPLE server solution.

Java SIP stack, originally based on mjsip.

PHP / ZeroC ICE based web administration system for Murmur Servers.

MumurWeb will be a comprehensive webinterface for Murmur servers based on PHP. Murmur is the server component for the Open-Source voice communication Mumble.

8ix Zenith CE spells Asterisk derived IP Telephony solution.When you want robust communication, without the huge cost and recurring bill that comes with the way you do business, 8ix Zenith CE addresses your every communication need.

Asterisk based callcenter system. Administration interface to manage queues, agents, accounts, skills - track calls, create statistics. Realtime queue monitor. Agent web interface with build in IAX softphone.

Twinkle is a VOIP based telephony software on linux. This project will install this software on MAC OS X using fink package management system. This will be included in stable tree of fink.

Queue-Tip has moved to Rubyforge - please go to http://queue-tip.rubyforge.org

CHAROZT is an Asterisk MOH (Music on Hold) Uploader. It is a Linux-based application installed in Linux desktops that converts MP3 to WAV for Asterisk supported MOH. It can also upload converted files to the Asterisk Server.

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.

Clock in/out from any phone, or from the web portal. Comes with full web portal and reports. Get the latest build from SVN Checkout (Public/SVN Repository). For more information on Asterisk PHP Timeclock, visit http://www.asterisktimeclock.org

Mongo has bundled Asterisk 1.4.13 + add-ons 1-4.4 + freePBX 2.3.1 + add-ons + sugarOS 4.5.1 + XAMPP 1.6.4 + MYSQL (with it's own sock) + PHP (part of XAMPP) + oscommerce

DiWA (Dialplan Web Administrator) is a php/ajax interactive web application to manage Asterisk's diaplan. It supports both static and realtime configuration, and doesn't require database.