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Gtk app that periodically checks AT&T Pay As You Go data.
Features:
Alerts on low balance, and expiration of airtime and feature packages.
Alert thresholds configurable by settings menu
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.
The Genalyze log analyzer, a simple yet highly extensible framework for parsing log files, was designed for parsing T-Server log files generated by Genesys contact center software platform, yet is generic and easy enough to adapt for other log formats.
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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).
Java Media Library for wiring together Sinks and Sources. Can have rtp Sinks and Sources, FileRecorders/FilePlayers, Microphone, etc. The framework has default sink/sources already. It does not provide Text-to-Speech Source or Speech Recognizer Sink.
A Sip live audio feeding agent. The agent captures audio from sound card and sends live audio stream(uLaw) to caller(sip phone) using RTP. It is based on Peers 0.3(http://peers.sourceforge.net/). Can be used in IP telephony to broadcast live audio.
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Clock in/out from any phone, or from the web portal. Comes with full web portal and reports. Get the latest build from SVN Checkout (Public/SVN Repository). For more information on Asterisk PHP Timeclock, visit http://www.asterisktimeclock.org
DiWA (Dialplan Web Administrator) is a php/ajax interactive web application to manage Asterisk's diaplan. It supports both static and realtime configuration, and doesn't require database.
Nhabla-Queue: Is a Asterisk report system for inbounds call center campaigns, provide many reports with useful information and graphics for analysis, which can be printed on PDF.
Using the new C# Dll from swig portability of Openobex/ObexFtp to write a nice GUI able to transfer file between two devices with obex protocol. More Info’s here: http://cs-obexftp.wiki.sourceforge.net/
Nhabla-Bill: Is a report system a billing application inspired on Asterisk-Stat by Areski, offering many useful reports for the analysis, add to the billing function.
Visual voicemail viewer for Asterisk/AsteriskNOW written in PHP. Users log in with their extension and v/m password and can download messages with the web browser.
J8051 is an API for communication between a 8051 microcontroller and a cell phone using Java ME, Bluetooth and "C" language callback functions, providing a easy way for embed systems developers build a communication between.
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
FreeRADIUS module providing connectivity to CDRTool prepaid engine. Provides prepaid authentication for calls proxied by OpenSER and returns to OpenSER MaxCallDuration and user Credit.
Can be easily extended to support other SIP or H323 devices.
Xen-Agi is an open source java library that allow you to build java applications that interact with the Asterisk Server for VOIP and PBX functionality. This project supports the FastAGI protocol exclusively and is inspired by the Asterisk-Java project.
The Jive API is a lightweight Java framework for writing IVR applications.
The premise for the Jive API is to allow powerful IVR applications to be written without the user having to worry (or know) about the underlying IVR platforms.
Java version of Dialogic's SwitchKit API. This API is actually a more "Java Centric" API, with listeners, filters, and a very extensible and modular framework that finally uses JNI to access the native libraries provided by Dialogic.
SIP to Skype Gateway/Bridge/Converter/Adapter. Make and receive SIP to Skype Calls and Skype to SIP Calls. Call Skype users using speed dial or use SkypeOut. Make SIP calls from Skype using a SIP provider or SIP PBX. Use as a Skype Trunk with a PBX.