Search Results for "debian gnu linux" - Page 3
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Showing 189 open source projects for "debian gnu linux"
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Custom VMs From 1 to 96 vCPUs With 99.95% UptimeLive migration and automatic failover keep workloads online through maintenance. One free e2-micro VM every month.
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skiputler makes your Topcom "Butler 4012 USB VoIP" phone work on Linux and Mac OS X. It's a userspace driver which connects the phone to Skype.
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Sippy B2BUA and RFC3261 SIP Stack
This project has been relocated to http://github.com/sippy/rtpproxy h
This project has been relocated to http://github.com/sippy/rtpproxy http://github.com/sippy/b2bua -
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Enhanced Callback
Adds enhancements to Freepbx callback script
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Artemisa is a honeypot for VoIP (SIP) networks. It is designed to connect to a VoIP domain as a user-agent backend in order to detect malicious activity at an early stage and also adjust the policies of the enterprise in real-time.
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mumb1e
A Murmur-Server Webinterface written in PHP for Mumble
Administration Tool for Mumble -/- Murmur Servers. Mumb1e Admin Plugin, also named "MAP" is a powerful Murmur web interface application written with PHP that can create useres, edit themself and a lot more for Mumble server. MAP is developed over m -
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ictbilling
A client management, billing and e-commerce solution
ICTBilling contains a set of Drupal 7 modules that can be used with Ubercart E-commerce solution and ICTBroadcast, a unified communication auto dialer software. This Client and Billing solution for ITSP has following modules and rules: client module, ictservice module, ratelist module and rules for ubercart. Client module handles clients user interfaces, subscribed service details, service links, list of available addons and current balance. Another module, ICTService creates simple... -
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Eolos VoipAudit
Universal Framework to audit Voip protocols's security
Visual Framework Tool to scan/sniff address space, enumerate users, crack credentials, pattern based dial spoofing and security reporting for Voip protocols. This software is intended to give a general framework to build and plug Voip protocol analizers in order to fix security issues and enhance VoiP platforms confidence. It IS NOT intended to be a cracking tool for malicious system breakers, but a really software MUST for security people to assure Voip platform security. -
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cdr2serial
Send Asterisk CDR to serial device
cdr2serial sends Asterisk CDR serial port for the use of software for hotels. They have been used libraries asterisk-java for agi events and jssc for send via serial port. -
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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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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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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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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. -
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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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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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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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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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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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agispeedy
The Agispeedy is an AGI Application Server implemention in asterisk
The Agispeedy is an AGI Application Server implemention in asterisk and written by Pure PHP. it inconceivable faster than normal agi about 1000%, and improves performance. -
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vtdialer
predictive dialing for vTiger using asterisk
This is a vTiger module adding predictive dialing features using the popular asterisk PBX. The module calls the leads included in active campaigns tuning the number of concurrent calls according with the number of available agents, the answer ratio and other parameters (dial time, conversation time etc.) It is included also an optional module providing statistics on the contact center behaviour -
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Cisco IP Phone Inventory Tool
Cisco IP Phone Inventory - Serial Number, MAC, Model, and more...
Project moved to github: https://github.com/vloschiavo/cipit
