Search Results for "set"

queXS is a web based, Open Source, CATI (Computer Assisted Telephone Interviewing) System. queXS integrates with queXML for creating questionnaires, LimeSurvey for collecting data and Asterisk for VoIP telephony.

Pyst consists of a set of interfaces and libraries to allow programming of Asterisk from python. The library currently supports AGI, AMI, and the parsing of Asterisk configuration files. The library also includes debugging facilities for AGI. 2014-04-17: Moved the version control to GIT. To check out see the tab "Code". Note that the whole history including ancient CVS, then some time in monotone, then subversion was united into one GIT repository thanks to ESR's reposurgeon. 2013-05-29: Maintainers of github fork "pyst2" contacted to join forces. ...

If you want to quickly set up a multi-tenant PBX on matter of few minutes, this is a suitable choice. - Demo environment. You can start to add extensions, gateways and dialplans, and quickly have a fully working PBX environment. - Import previous FusionPBX backup to Live environment. After this you will have a quickly working PBX on few minutes. - Install on hard disk.

VoIP Honey project provides a set of tools for building an entire honeynet, thus includes honeywall and honeypot emulating VoIP environments such as Asterisk PBX or OpenSer with fully configurable connections. Voip Honey runs on GNU/Linux and Windows Systems. It can be compiled for Mac OSX as well.

Sipp is a performance testing tool for the SIP protocol. Its main features are basic SIPStone scenarios, TCP/UDP transport, customizable (XML-based) scenarios, dynamic adjustment of call-rate and a comprehensive set of real-time statistics.

The PoC telephony application suite is a set of scripts and libraries for interactive telephony environments such as the Asterisk PBX and Yate.

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

Session Initialtion Protocol (SIP) is widely used for telephone services over the Internet. CafeSip provides a suite of open-source tools and applications for creating customized SIP services and applications using the Java.

The Asterisk-Java package consists of a set of Java classes that allow you to easily build Java applications that interact with an Asterisk PBX Server. It supports the FastAGI protocol and the Manager API.

GSM Phone manager with a friendly graphical user interface. It uses QT for GUI, Bluetooth technology for connection and the GSM extended set of AT commands for the dialog with the device.

rtusd is a foundation set of real time enabled User Space Devices/Drivers designed to link to HRT high resolution Timer controlled POSIX threads and link re-factored Linux hdw drivers for 2.6.18+. rtusd propogates Linux RT to user space generically.

A set of JNI classes providing direct access to Asterisk PBX functionality from Java. This is not a socket-level interface into the Asterisk manager app but a true Java<->Asterisk integration at the Thread level.

comtorJ2ME is a library that simplifies J2ME software development. This is a set templates (java interfaces) that encapsulates Form , List and Canvas frecuent initilizations. Includes other useful classes RMS , StringTokenizer ,HttpConnection

A set of perl scripts which senses wether a particular bluetooth device is within range, also featuring sync with IrMC 1.1 compliant devices.

pblogan (Phone Box LOG ANalyzer) A set of perl scripts that extract information from a Panasonic Phone Box via serial port, and upload them to a Postgres Database which is queried by several php pages.

tprsh (Twisted Perl Remote Shell) is a set of client/server Perl scripts which allow the server to execute commands on the client via TCP. The scripts use the IO::Socket module and run on Linux and Windows.

A set of classes and tools aimed at making it easier to develop MIDlet games (J2ME MIDP 2, for cellullars, PDA:s...) Classes include extended sprites and maps, tools will be a mapeditor and a spriteanimator.

Call accounting/billing software which uses SMDR facility of phone centrals. It uses a set of scripts for data(calls) aquisition/recording and preprocessing before import into a mysql database. The billing is done with PHP using a strong set of criterias

Cisco voice web interface is a web tool designed to help maintaining VoIP on cisco platforms. Based on a set of perl parser's it now allow's to perform quite all operation's needed to check voice call's. In close future planned to add dial-peer crea

Open Telephony Server or OtServer for short it's a set of tools for developing telephony applications. A CallFlow Graphical Editor. Dialogic support.

The asterisk-jsp package consists of a set of jsp tags and a tomcat connector that allow you to easily build jsp applications that interact with an Asterisk PBX Server. It's using asterisk-java as the back end to do the heavy lifting.