Search Results for "gnu/linux" - Page 5
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Showing 308 open source projects for "gnu/linux"
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OpenVOIP
Padrão de comunicação VOIP.
O OpenVOIP é uma solução Open Source para que empresas possam ter um padrão de comunicação VOIP. -
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YaaCs - Yet Another Advanced Cati system
survey research software integrate for CATI/ CAWI/ CAMI survey
YaaCs, developed by Demetra www.opinioni.net, survey research via phone CATI (also Voip) and web CAWI and for telemarketing. Develop questionnaire for survey, create probabilistic sample, monitoring Interviewer, export survey data in statistical.. It's integrate with www.modsurvey.org. Sviluppato da Demetra www.opinioni.net per indagini CATI/ CAWI e CAMI. Vi permette di programmare dei questionari elettronici, campionamenti, monitorare gli intervistatori, esportare dati in formati... -
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JADfactory
Application to make JAD files
A simple application to make jad files -
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Fix Skype
Fix Skype fixes future timestamps in the Skype messages database.
Fix Skype is a python script and Win32 binary that is used to fix "future" timestamps in the Skype message database, such as those caused by the end of DST or by dual-booting a Linux system with time set to UTC from anywhere west of GMT. Essentially, you feed it your username and the number of hours by which some timestamps are in the future, and it subtracts that amount of time from any timestamp greater than "now". -
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KAMAILIO (OpenSER) - robust, secure and scalable Open Source (GPL) SIP (RFC3261) server implementation with large features set (over 90 extension modules). As of May 2009, source code is hosted by GIT repository at http://sip-router.org
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SMPTE 2022-1 library for VLC & co
Optimized and cross platform SMPTE 2022 FEC library in C, Python, Java
Project moved to GitHub. https://github.com/davidfischer-ch/smpte2022lib -
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VoIPGMap
Graph active VoIP calls on Google Maps.
A bash shell script which will read active call destinations in E.164 format from a source (such as Asterisk), then run the called number through some plain text and MySQL databases to find the best latitude/longitude match. It then creates a .html file with Google Maps API JavaScript code which will display a single call origination point and the active destinations (calls) on a Google Maps world map. -
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astercc is a realtime billing solution for asterisk, could work with all kinds of asterisk based system and no need do any change to your original system. astercc could be used for hosted callshop solution, pbx billing solution.
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Root Lounge - Mumble Skin
A Skin for Mumble
Spice up your Mumble experience using this recreation of 4 different skins merged and adjusted into a new one. It is tested and running stable using Mumble 1.2.x. To install: Download, extract and follow the readme.txt contained in the archive. All credits go to the devs of the individual pieces including Mumble itself. -
AI-powered service management for IT and enterprise teamsGive your IT, operations, and business teams the ability to deliver exceptional services—without the complexity. Maximize operational efficiency with refreshingly simple, AI-powered Freshservice.
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An open source software project delivering converged voice and data applications. The StarPound platform enables you to rapidly develop voice and data solutions using the power of business process modeling. More info: http://www.starpound.org.
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Open Source ITSP / Hosted PBX software that sits on top of Asterisk to handle customer management, rates via LCR, cdr reports, multiple phones per customer, customer portal, among many other things.
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Turn.Message is .NET parser / generator of the TURN (Traversal Using Relay NAT) messages written in C#. Compatible with [MS-TURN] extension and clients based on Microsoft UCCAPI SDK.
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Turn.Server is .NET TURN (Traversal Using Relay NAT) server written in C#. Compatible with [MS-TURN] extension and clients based on Microsoft UCCAPI SDK.
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OZEKI VoIP Service Checker
The OZEKI VoIP Service Checker allows you to check VoIP services.
The Ozeki VoIP Service Checker is a tool you can use for checking the availability of VoIP services. It can be used to verify if a give service exists, whether it can be reached from a particular network and it checks if the user details, such as the username and password to access that service is correct. This tool was developed because many mobile Internet providers block VoIP traffic and no other tool exists that makes it possible to quickly verify if a certain network support... -
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asterCRM is a call center software for asterisk based VoIP system, also it has some CRM functions. It provide useful features such as pop-up, predictive dialer, click to call, extension status .... astercrm could work with all asterisk based system.
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MobilIVR is essentially a convenience layer that sits on-top of Asterisk [www.asterisk.org], as well as Kannel (SMS gateway).
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RightClickLync
Adds right click context menu to call a highlighted number in Lync
Adds a Right Click context menu to your browser to call number from Lync. Highlight a piece of text in IE/FF13/Chrome and the script will strip out non numerics and pass to Lync via tel: handler and Lync can normalise the number. -
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ACR - Asterisk CDR Reporter
Information on Asterisk cdr like ASR, ACD, Billing duration 'billsec',
Information on Asterisk cdr like ASR, ACD, Billing duration 'billsec', time criteria not only date, I will add more features once I have time, like channels usage and time zone shift. you are welcome to suggest ! Please feel free to download, install and/or modify the code and let me know your feedback. You can use the following CLI commands to download and install the script: cd /var/www/html wget http://citylan.dl.sourceforge.net/project/acrasteriskcdrr/rpt.tar.gz tar xzvf... -
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Hanasu
A P2P Messenger which uses JXTA. It provides secure RSA/AES end-to-end
Does not work because there is no master server! A P2P Messenger which uses JXTA. It provides secure RSA/AES end-to-end encryption and VoIP-Capabilities. -
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LuaSofia is a Lua binding of Sofia-Sip library. Sofia-SIP is an open-source SIP User-Agent library, compliant with the IETF RFC3261 specification. We decided to use git, source code can be get at: git://github.com/ppizarro/luasofia.git
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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
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wimovo
Client VoIP for Windows Mobile
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A simple interface to work with SIP numbers of Asterisk. Add, delete, and edit SIP numbers in the MySQL database As well as viewing CDR records Простой интерфейс для работы с SIP номерами сервера Asterisk. Добавление, удаление и редактирование SIP номеров в базе MySQL А так же просмотр CDR записей
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ChatAnt
client-server full java chat audio-video file sharing
client-server full-java chat, provided with audio&video (JMF), custom rooms, custom forums, instant- messaging and file sharing -
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CafeSip - Look what Java and SIP can do
A suite of open-source tools and frameworks for creating SIP apps
Session Initialtion Protocol (SIP) is widely used for telephone services over the Internet. CafeSip provides a suite of open-source tools and applications for creating customized SIP services and applications using the Java.
