Open Source ChromeOS VoIP Software

Elastix is a software-based PBX powered by 3CX and based on Debian. An open-standards solution, Elastix is an easy to install and manage UC system compatible with popular IP phones, gateways and SIP trunks. Elastix is complete with unified communications features such as integrated WebRTC video conferencing, chat, presence and softphones and smartphone clients for Windows, Mac, iOS and Android.

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 for a either a sound input(microphone) or sound output(speakers). Can also be used to gate noise from your own mic or play your microphone through your speakers. REQUIREMENTS: - Java 7 or higher for Windows. - Java 6 or higher for Mac. Java 7 recommended. - A virtual audio cable is required for use with VOIPs: For Windows users I recommend the VB-Cable driver (http://vb-audio.pagesperso-orange.fr/Cable/index.htm). Mac users can use SoundFlower.

Open Phone Abstraction Library (OPAL) is a C++ multi-platform, multi-protocol library for Fax, Video & Voice over IP and other networks. Also included is the Portable Tool Library (PTLib) which is a C++ multi-platform abstraction library and collection o

The Mumble PHP Interface (MumPI for short) allows you to manage your Mumble servers via a webinterface. Users can register, upload textures, and retrieve account data. Admins can manage virtual servers, registrations, admin accounts, online users…

Project moved to github: https://github.com/vloschiavo/cipit

FreePBX (formerly Asterisk Management Portal) is a project to bring together best-of-breed applications to produce a standardized implementation of Asterisk complete with web-based administrative interface.

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

Baresip is a portable and modular SIP User-Agent with audio and video support. the latest source code can be found here: https://github.com/alfredh/baresip

jPBXLite is a VoIP/SIP PBX. Supports SIP extensions, voicemail, trunks, conferences, queues (ACD) and an IVR system. Support video conferencing with jPhoneLite/1.4.0. NOTE:THIS PROJECT WAS RENAMED AND IS NOW jfPBX. Please go to jfpbx.sourceforge.net

Google Contacts to Grandstream xml Phonebook Format Converter

VoIP/SIP AutoDialer. Broadcasts a message with the option to transfer person to another number. Now includes new survey options.

A Chrome extension that turns the browser into a call centre agent console using WebRTC, so agents do not need a desk phone or a separate softphone running alongside their CRM. It registers as a SIP endpoint over WebRTC and puts the call controls where the agent is already working. Answer, hold, transfer and hang up happen in the browser, next to the customer record rather than in another window. Built to work with our contact centre products, but it speaks standard SIP over WebSocket, so it is usable against other platforms that do the same. GPL-3.0 licensed.

This is a web based instant 1 on 1 private online video conferencing solution. It's a solution for conducting easy to setup face to face meetings without leaving your office or home. It's the easiest and most cost-effective way to meet somebody and discuss one on one, to make a video call just by providing a private room access link.

This is a simple tool to map information for the cost of phone calls given by your VoIP provider, to the call log inside your Asterisk. As a result, you will know how much each of your employees has spent for phone calls. Additionally, this tool shows information about the assigned DID and CLIP numbers in Asterisk. This is helpful to identify obsolete, duplicate and multiple assignments.

Astmontray is a systray applet written in PyQt so users can monitor whether their PABX servers running Asterisk.org, the open source telephony platform, are up or not. It supports AMI authentication and runs fine on Linux, OSX and Maemo tablets.

FATS - FATS is a Twisted and Fast Asterisk's Telephony Services. Project contains implementation of FastAGI, AMI protocols for the Twisted framework. Using it you can develop fast and pretty services for the Asterisk IP-PBX.

This is a script to engage telemarketers and waste their time. Also records the conversation for listening later.

This is a repository for QueueMetrics add-ons that are contribuited by the community and are open-source.

Plug and play provisioning server for snom VoIP phones written in Python. The server supports the plug and play provisioning feature embedded in snom phones of the snom 3xx, snom8xx and snom MeetingPoint series.

Twisted Protocols for communication with FreeSWITCH PySWITCH is a Twisted Python library that allows you to communicate with FreeSWITCH using inbound and outbound EventSocket connections.

A complete contact center solution for the Asterisk PBX, capable of handling inbound and outbound campaigns. Provides an agent toolbar for CTI. For managers an admin panel controls every aspect of the call center. Stunning historical and realtime stats.

AS3 skype API for the Skype. Flash has access to the system Webcam, Audio and flash 9+ has all the capabilities to build a Socket communications. So we can have a Flash based skype client on the web. This can be a widget for instant chat & voice call.

Cross-platform Java Teamspeak client

The Asterisk Config PHP-Parser claims to be a simple but effective function writen in PHP non-OOP that is capable to parse any standard .conf Asterisk configuration file an drop the data into a multi-dimensional array.

Asterpoll is an addon to asterisk that allows a asterisk admin to create a poll question or questions . this poll is a destination on the PBX. callers answers are recorded and the results are shown in a graphical environment.