Difference between revisions of "Category/Business/telephony"
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Revision as of 15:53, 23 January 2015
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- ADM (Asterisk Desktop Manager) aims to integrate your desktop with the Asterisk PBX and hardware IP phone by providing some useful features such as automatic on-call volume reduction, one click dialling (from the clipboard), automatic messenger "away" status when on a call, and some Cisco 79xx specific features. It runs with any GNOME compatible window manager.
- ANT is a desktop ISDN telephony application written for GNU/Linux. It supports OSS (Open Sound System) and I4L (ISDN4Linux). Its user interface was made for GTK+ 2.x (GIMP toolkit). It directly interfaces OSS and ISDN devices, so there is no need to install extra software or hardware like PBX or telephony cards, if you've got direct access to an audio capable ISDN card and a full duplex soundcard or two sound devices.
- The ATSlog software provides a handy web-oriented interface for viewing and analysing calls for various types of PBX (Private Branch eXchange) models.
At present the program operates successfully with Panasonic, Samsung, Hybrex, Siemens, LG, and Alcatel PBX models. If your particular type of PBX is not supported yet, we can add this functionality.
- Asterisk is a complete PBX in software. It does voice over IP in three protocols, and interoperates with almost all standards-based telephony equipment using relatively inexpensive hardware. Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, and Call Queueing. It also supports three-way calling, caller ID services, ADSI, MGCP, SIP and H.323 (as both client and gateway). The system needs no additional hardware for Voice over IP. Asterisk connects with digital and analog telephony equipment through a number of hardware devices, including that manufactured by Asterisk's sponsors, Digium. Digium has single and quad span T1 and E1 interfaces for interconnection to PRI lines and channel banks. An analog FXO card is available, and more analog interfaces are pending.
- Asterisk Manager Suite
- AMS (Asterisk Manager Suite) is a suite of software intended to make day to day administration and monitoring of an Asterisk PBX server easier. It contains a daemon that acts as a proxy to Asterisk's Manger Interface and a GTK GUI application for monitoring and administration.
- Asterisk Managment Portal
- * AMP is now "FreePBX" The goal of the Asterisk Management Portal (AMP) project is to bring together best-of-breed applications to produce a standardized implementation of Asterisk complete with a Web-based administrative interface.
- 'asterisk-oh323' adds H.323 support to the ASTERISK soft PBX by interfacing the OpenH323 library to ASTERISK through a loadable module. The package provides the channel driver as well as a wrapper in a shared library form. It can initiate and receive calls to and from H.323 endpoints, and has been successfully tested with the H.323 terminals (ohphone, openphone) on the OpenH323 site.
- Bayonne is the telephony server of the GNU project. Based on the ACS project, it offers a multi-line interactive voice response telephony server which may be scripted and telephony plug-ins for runtime driver configuration directly extended thru modular plugins. Bayonne also features "TGI" for making Perl applications "telephony aware". Support has been extended to include XML parsing and support has been started on VoIP integration to support next generation telephone networks. The project is not fully completed but is moving steadily towards producing a finished project that may be used to build telephony based system administration, home automation, automated attendant, v-commerce, and voice messaging systems.
- GNU ccAudio is a stand-alone C++ class library and newly designated GNU package for manipulating audio data, whether on disk or in memory. GNU ccAudio offers the ability to work with audio file formats on disk by treating audio data as sequenced arrays of sample data rather than as arbitrary octets as some audio file manipulation libraries do. In addition to being audio content aware, GNU ccAudio allows header manipulation for setting things like annotation fields. GNU ccAudio is also endian aware and highly portable to both posix and win32 based systems. GNU ccAudio also offers basic audio signal processing including tone data set generation and pluggable codec operations. In the future we will provide loadable free software audio codec modules for many common audio encoding formats where not patent encumbered.
- GNU ccRTP is a high performance threadsafe C++ RTP (Real-Time Transport Protocol) stack. It can be used to build both client and server applications for audio and visual conferencing over the Internet, for streaming of realtime data, and for next generation IP based telephony systems.
The XiVO Distro is an open source unified communications software system:
- The graphical user interface (GUI) provides for configuring, controlling, and managing the IP PBX software.
- The XiVO Distro includes packages that offer this features: VoIP, PBX, Fax, voice-mail and email functions.
- XiVO is licensed under the GNU General Public License V3 (GPL V3). XiVO is based on the free software Asterisk and the XiVO Linux distribution is based on Debian Wheezy.
- The official webpage: http://xivo.io
- The documentation: http://documentation.xivo.io
- On Github: https://github.com/xivo-pbx/xivo-doc
- Projects, roadmap, bug tracking: http://projects.xivo.io/projects/xivo/boards
- Blog: http://blog.xivo.fr
- IRC: server freenode, channel #xivo
- Asterisk et XiVO: http://www.asteriskexchange.com/listings/609
- Try the software: http://www.xivo.io/#section11
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