Search Results for "simple-xml"

Teleweb is a simple but very efficient call center software. It is the only solution out there that is completely wireless and hosted online. No need for landline, softphone or Voip. All you need is your computer, a headset and you can call the world online.

...By leveraging the Application API, third-party applications can perform commands such as starting an audio or video call, answering a call or placing a call on hold. CounterPath is now encouraging third-party-application developers to integrate their applications with Bria. Suggested applications range from simple functions such as enabling click-to-call from web pages, to integration with sophisticated Call Center, Customer Relationship Management (CRM) and Health Care applications. This project provides sample and convenient wrapper code to utilize this API. *Note* you need a Bria desktop application to utilize the API.

This project implements a simple STUN server and client on Windows, Linux, and Solaris. The STUN protocol (Simple Traversal of UDP through NATs) is described in the IETF RFC 3489, available at http://www.ietf.org/rfc/rfc3489.txt

Voxy - the simplest way to integrate VoiceXML applications in Asterisk. Configure your dial plan with the URL of your VoiceXML application and it's done.

Peers is a very simple softphone. It's a SIP User-Agent, written in java, it works on windows, linux and mac. It can be used with SIP servers like opensips or asterisk IPBX. It supports G711 codec (PCMU and PCMA) and telephone-events (DTMF).

This is a P2P VoIP system. It support audio and video chat through kademlia-based dht overlay. The p2p module use Kademlia protocols; Audio codec is G.729. An simple Introduction here: http://www.needdo.com/2007/09/ppphonekademliap2p-voip/ . My Personal blog is http://www.idolf.cn and my email is dolfcao@gmail.com .

...The SIP clients make use of Microsoft UCC API SDK, ensuring the highest quality of audio and video. Compatible with Office Communications Server. See also open source, cross-platform: 1) simple messenger Brief Msg at http://briefmsg.org 2) MUVConf is a multi-user video conferencing, see demo video http://youtu.be/YrBU-Aqtvrk, download https://code.google.com/p/muvconf/downloads/list

A Easy 3G dialer. A dial system with support for most current GPRS and 3G phones. Simple, the idea is to detect, connect, use and disconnect . Languages: pyQT4, Shell Script.

this is to create an chatting module which is easy to implement

VMukti open source is leading Asterisk/ Yate enabled p2p Video IP Communications Suite for Web / PSTN. These serverless broadband ready platform enable OS community to save 90% on capital & operating costs over proprietary software for conferencing &

Skype Audio Player allows playing audio files (mp3, wav, wma) during Skype calls to both parties. It is a simple tool without luxurious functions, useful for those who study foreign languages using Skype and need to play sample dialogs or audio tests.

Kiax is a softphone (soft phone, VoIP client) with a simple and comfortable user interface for making VoIP calls to Asterisk PBX. It depends on the iaxclient library to use Asterisk's IAX2 protocol for easy call configuration and audio setup.

It's a teamspeak viewer based on "Teamspeak Viewer", but made in AJAX the XML is generated with PHP the display is done with javascript

This is a general-purpose library for communicating with the Skype API. It is simple, cross-platform, and easy to use. The library currently works on Windows (Win32) and Linux (X11/D-BUS).

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.

Simple, small, secure web conferencing on any platform that supports Java.

Softphone SIP com voz, vídeo e Mensagem Instantânea. DTMF, Nat-Traversal, Suporte a Skin(Macromedia Flash), SMS, Historico de Chamadas, Presença, Hold, Auto-Answer e etc. Sip Softphone with Voice, Video and IM.

SIPStir is a bulk call generator for SIP devices/servers. The server piece is capable of generating an infinite variety of call scenarios based on a simple scripting language. A wxPython client can connect to the server to monitor and modify progress.

The RTP Monitor discovers RTP flows, used by Internet phone applications, and monitors their QoS metrics, observing the interference caused by traffic congestion. It uses distributed collecting agents and provides feedback about the quality in the calls.

The Cisco IP Telephony Services project contains many IP phone services and utilities for the Cisco 79XX phones and CallManager. This project houses several scripts, services, and administration tools. Recruiting! Contact me and become a developer.

This simple php page show a list of queues and his members and agentes activity by quering Astrerisk through the Manager Interface, reloading itself every n seconds. Suggestions are welcome.

SOAPMODE : A portal for Imode (TM DOCOMO) client, using web services (SOAP).

Open79XX XMLDirectory is an open PHP-based XML menuing system for providing on-screen services to the Cisco 79XX Series IP Phones. It contains dynamically generated phone directories, on-screen memos, links to other services, and many more options.

Utilities enhance AMP, A@H, Asterisk and Cisco XML. Speed Dialing revised using web interface. Paging using custom meetme routines. Intercom by dialing 0 + Ext #. Browsing AMP's extensions on your phones LCD. Backup patch for Asterisk@Home maint screen.

OpenIM is a instant messenger system utilising the XMPP (Internet-standard communication protocol) and is built on the Java platform. It supports many popular systems (MSN, ICQ, AIM, Jabber, Google Talk etc.) and supports Voice (and Video) Over IP.