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ICTAgent is a advance webphone offered as chrome extesion and when installed it will convert all available contacts as click to dial. This extension will embed WebRTC based VoIP Phone in browser which will remain connected to the main server, and will be responsible for inbound and outbound calls, transfer calls, sending DTMF as well sending Fax. It can harvest contacts from the random pages. Agent can access contents, contacts from a single application. Browser can automatically fetch required URL depending on the course of call. Automatic URL will eliminate any delay which agent take while searching and finding desired data. ...

JAVIER is a JAvascript Voicexml InterpretER, designed (but not restricted) to run inside a web browser, its main engine has less than 1000 lines of code. It's maybe, the tiniest but (almost) FULL VoiceXML implementation.

A Mod so that Minecraft now natively supports Mumble's positional audio feature. This means: Directional and positionally attenuated VOIP in relation to the game world. Please visit the Forum for information about the newest Version! Main Forum-Thread: http://www.minecraftforum.net/forums/mapping-and-modding/minecraft-mods/1272675 TheSkorm's Fork on GitHub: https://github.com/TheSkorm/mod_mumblelink Mumble: http://mumble.sourceforge.net Donations are greatly appreciated: https://sourceforge.net/donate/?user_id=2370023

This a Grapical User Interface for Gammu - a free GSM library. The export of phone data via iCalendar is main goal.

Asterisk PBX Integration Zimlet is an Extension for Zimbra Collaboration Suite. The Zimlet does Interface with the Asterisk Manager Interface to integrate with Asterisk PBX. The main focus is dial-on-click for Phone numbers inside Contacts and Emails.

This project is developed as a database application using Qt 4.0 and MySQL 5.0. Main aim of the project is store call details of each customer. At any time (usually end of month) admin can send call details to the selected customer.

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 would like to say thanks to all the people that in some way contributed with the project. ...

Experimental open source project, which aims to test possibility of VoIP calls "from browser to browser". Main part is Java application implementing UDP hole punching, which is sent to both clients using JWS. See it online in action at project homepage

TRAUMA helps you to analyze your telephone bills of the Deutsche Telekom (main German telephony provider). | TRAUMA wertet Telefonrechnungen der Deutschen Telekom aus, wie man sie unter RechnungOnline als SYLK-Tabellendatei herunterladen kann.

SMSvm can send many messages by using HTTP protocol and Mail-Agent Protocol. But the main idea of this program is send SMS without any limits. The cost of this is typing those digits per each account :) Sorry, the source was posted in forum :) Look there

GAEM stands for Graphical Adventure tiny-Engine for Mobiles. Its main use is to read some text files and images and generate the game. It's being developed in J2ME under CLDC/MIDP environment.

Bogotel simulates Intel Dialogic hardware for programs using Dialogic APIs. It allows programs that use the System Release 5.1.1 to run without special hardware (or the SR being installed). Includes support for main Global Call and some voice features.

...Sippet is an open-source SIP User-Agent library, compliant with the IETF RFC 3261 specification. It can be used as a building block for SIP client software for uses such as VoIP, IM, and many other real-time and P2P communication services. The main target was to enable Javascript applications to use UDP, TCP and TLS transports along WebSocket. Existing SIP solutions for the browser are forced to use the WebSockets API to send/receive SIP messages. But for many service providers, the WebSocket protocol could not be a feasible solution due to scalability, support and other business constraints. ...