Search Results for "non-free"
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Showing 40 open source projects for "non-free"
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Siproxd is a proxy/masquerading daemon for the SIP protocol. It allows SIP clients (softphones & hardphones) to work behind an IP masquerading firewall or router.
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VitalPBX
Unified Communications System - PBX System
VitalPBX is a free telephone and communications PBX system for companies. It is a complete platform that can be installed on the physical hardware on the site or as a hosted application. VitalPBX acts as the upper layer interface for the Linux base and then Asterisk (one of the most popular communication toolkits in the world). For this reason, VitalPBX is the graphic user interface between you and the complex world of modern communications. -
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Free Telephony Project
Free software and hardware for telephony
Legacy SVN repositories for projects from David Rowe http://rowetel.com Mailing lists for Codec 2 and SM1000. -
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OpenSIPS (former OpenSER) is an GPL implementation of a multi-functionality SIP Server that targets to deliver a high-level technical solution (performance, security and quality) to be used in professional SIP server platforms. IMPORTANT: this is no longer the main hosting for the project. This was moved on GITHUB - https://github.com/OpenSIPS/opensips
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Ship Agents FasterGemini Enterprise Agent Platform lets you rapidly build, scale, govern and optimize production-ready agents grounded in your organization's data. The platform enables developers to build custom or pre-built agents for virtually any use case. New customers get $300 in free credits.
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SIPSL (SIP Service Layer) is a Programmable Session Border Controller (SBC) and SIP B2BUA, as defined in RFC3261. Written in c++ and multithreaded. The Application logic can be implemented by extending the application class and implementing the call backs.
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GNU Gatekeeper (GnuGk)
H.323 Gatekeeper for VoIP and videconferencing
The project has moved! Please find current versions at https://www.gnugk.org/ The GNU Gatekeeper (GnuGk) is a full featured H.323 gatekeeper under GPL license. It supports VoIP and videoconferencing and includes Radius and database support and many call routing functions. The project has moved! Please find current versions at https://www.gnugk.org/ -
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TurnServer is a implementation of Traversal Using Relay around NAT (TURN) protocol. This protocol allows a client to obtain IP addresses and ports from such a relay.
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PBXlab adds value to business thanks to its valuable contribution of technological solutions. PBXlab promotes the use of free software programs that can be tailored to the real needs of its users, speed of movement between users, at a very low cost or no cost. PBXlab offers development of add-ons and improvements to programs, leading to more full, safe and adapted to customer needs programs. We offer technological independence without having to develop products from scratch. ...
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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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VoIP Honey project provides a set of tools for building an entire honeynet, thus includes honeywall and honeypot emulating VoIP environments such as Asterisk PBX or OpenSer with fully configurable connections. Voip Honey runs on GNU/Linux and Windows Systems. It can be compiled for Mac OSX as well.
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AVAYA CMS API Consumer samples
API Consumer of AVAYA CMS reports
Free .NET 4.5 consumer of AVREMO.CMS.API.LimitedEdition AVREMO.CMS.API is able to Show AVAYA CMS reports as HTML, JSON and XML Render. It includes also a wrapper : .NET Client API (AVREMO.CMS.Contracts.dll) enabling fast development in .NET 4.5 consumer applications. AVREMO.CMS.API.FullEdition download page: https://sourceforge.net/projects/avayacmsapiserver/ -
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AVAYA CMS API Server
CMS API Server with web site
Free version of AVREMO.CMS.API exposing REST Services. With this software you can browse for all CMS reports. Design a report with AVAYA CMS Supervisor designer, application automatically expose data as XML or JSON. You can also browse reports with your preferred browser (desktop or mobile) and execute CMS operations like modify dictionaries, modify agent skills, modify VDN/Vector associations and more. -
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NoiseGator (Noise Gate)
A simple noise gate app intended for use with VOIPs like Skype.
Ever wanted to cut out background noise when talking with others on Skype? Now it's possible! NoiseGator is a light-weight noise gate application that routes audio through an audio input to an audio output. In real-time the audio level is analysed and if the average level is higher than the threshold the audio bypasses as normal. However, if the average level goes below the threshold, the gate closes and the audio is cut. When used with a virtual audio cable it can act as a noise gate... -
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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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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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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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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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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 rpt.tar.gz then point your internet browser to http://your-ip-address/cdr.htm -
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node (nodejs) non-blocking RADIUS client wrapper
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VOIP client/server in python >= 2.6. Audio in/out: ossaudiodev (UNIX like) or SoX. Network: bzip2 compression, speex or ogg audio compression, you can configure all, minimum bytes per second: 350-400 in speex mode: U8, 6 kHz, quality 0, bzip2, buf 4K
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This project is aim to develop module named app_avi.so for asterisk to add the ability of play avi file.
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The goal of the project is to create a high-performance, open-source and standards-compliant implementation of a Home-Subscriber-Server (HSS) for use in a IMS context.
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[es ] Aplicación web central para particulares. [en ] Core web application for individuals. [fr] Application Web centrale pour les personnes. [pt] Central de aplicação web para indivíduos. [ar] المركزي تطبيق ويب للأفراد
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Simu server project is aimed for IP telephony (VoIP) between classical SIP phones and Flash streaming applications.
