Search Results for "phone"

Kablink open team collaboration software uses social networking to unify team workspaces w/ real-time web conferencing. Collaboration for knowledge networking, program management, communities-of-practice, telework, other business process/functional areas

A C++ framework utilizing Design Patterns for creating Linux and Windows communications applications that contain Dialogic® products. Includes media and network classes (analog, digital, SIP, H323), multithreaded event handling, distributed app support.

Replacement for the SCCP channel driver in Asterisk. Extended features include Shared Lines, Presence / BLF, customizable Feature Buttons, and Custom Device State. Visit our discussion mailing list for help and join us as a developer if you like. The project moved to https://github.com/chan-sccp/chan-sccp

Add the possibility to search and call endpoints from Cisco Meeting Management or from Cisco Endpoints (SX10/SX20/Codec Plus/Spark Room/Etc) without having a Cisco TMS (TelePresence Management Suite) server. Cisco didn't include a phone-book or the possibility to create a new video-conference from CMM. Those functions are still in the old TMS server where you need to buy a license for every endpoint you add... Anyway, this tool emulates the TMS PhoneBook and allows you to search and add new participants to a existing conference. The search is regular expression and the filter compare the user query with the endpoint name or H323/SIP dial string from the database. ...

Install your own Free Voice Chat Party Line that runs 24 hours a day! No phone numbers to call, a complete mobile friendly web based Party Line Script. Using the latest browser technology to deliver live free voice chat from pc's and cell phones to any HTTPS web page in the world. Works on any cheap shared Linux server hosting platform with HTTPS. Can be used for Professional Business Owners support and information pages.

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/

wxCommunicator is a cross platform SIP softphone written in C++ utilizing customized sipXtapi user agent library and wxWidgets 2.8.9 GUI library. For a list of supported features see http://wxcommunicator.sourceforge.net/features.html .

Konference is (or better: will be) a video-conferencing application for KDE. Since the rewrite (2005/01/25) it supports SIP as the signalling protocol. No longer H323 folks.

dtmfbox is a small softswitch application (SIP/CAPI), that can be used to control different tasks over telephone keyboard via DTMF. Mostly, it was made to run on the AVM FRITZ!Box 7170 (mipsel) but works under Unix/Linux and Windows, too.

This project aim to provide a web based application to plan Conference Call using Asterisk application MeetMe(). The engine is based on Vaadin language to improve the portability of the application, Hibernate and some AGI script based on PHP.

Mission Control is a Telepathy component implementing the Account Manager and Channel Dispatcher specifications. It stores IM account settings, brings accounts online as necessary, and hands communication channels to the relevant applications.

Sofia-SIP - a RFC3261 compliant SIP User-Agent library.

Google Contacts to Grandstream xml Phonebook Format Converter

WebPhone Mural Chat é um projeto que realiza comunicação (chat) entre uma página Web e um Celular. Ambos conectam à um banco de dados MySQL, que provê e armazena os dados do chat, utilizando-se apenas de uma Rede Wireless. O Workflow é Twitter

A Client-Server application to make votings with life evaluation via Bluetooth. Server is written in Java for use an any platform. Client is a Java-midlet for use on most mobile phones. It is result of a TU-Ilmenau software project. (See the Credits)

LibDreamSpeak is an example implementation of the TeamSpeak2 protocol (client-side) in Java. Some day it may be used as base for alternative TeamSpeak2 (mobile?) clients.

8ix Zenith CE spells Asterisk derived IP Telephony solution.When you want robust communication, without the huge cost and recurring bill that comes with the way you do business, 8ix Zenith CE addresses your every communication need.

Queue-Tip has moved to Rubyforge - please go to http://queue-tip.rubyforge.org

Prototype testbed implementation of the IETF Media Server Control (MEDIACTRL) SIP Control Framework, comprehensive of both control and processing functionality (as in IMS MRF, Media Resource Function).

RTP text/t140 Library is a reference implementation for RTP Payload Type for Text Conversation (RFC 4103). The library has source code for encoding and decoding RFC 4103 data, and may be used either as a plug-in to JMF or in a separate RTP sender/receive

Chabber is an extensible XMPP-based Instant Messaging application. It is a generic, portable application that follows the standard protocols for Instant Messaging and gives users a lot of advanced functionalities. It provides a platform for plugins.

A lightweight cross platform IP telephony client using the IAX protocol, designed for use with the asterisk open source PBX.

IHU is a VoIP application for Linux (using Qt and Speex), with low latency, crypted stream, minimal use of bandwith, and without intermediary servers. It is the easiest way to talk real-time with your friends (like phone) on the internet or LAN.

CONFIANCE stands for CONFerencing IMS-enabled Architecture for Next-generation Communication Experience: an implementation of the IETF XCON (Centralized Conferencing) framework and of the BFCP (Binary Floor Control Protocol)