Free Software Foundation!

ADM 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 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.

ATSLog 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 2 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 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 '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 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.

CcAudio2 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.

Ccrtp 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.

Ccscript GNU ccScript is a C++ class framework for creating a virtual machine execution system for use with and as a scripting/assembler language for state-transition driven realtime systems. It is the core of the scripting engine found in GNU Bayonne. It is meant to be used where step execution is important, and where each step is in response to a callback event or a state machine transition. It offers deterministic execution and low overhead so that many concurrent instances can run together. However, in addition to offering step machine execution, GNU ccScript loads all scripts into an active image at once. This is for performance, as all operations in the script system, to assure deterministic execution, are in memory. GNU ccScript also offers the ability to load new scripts en masse. Existing active sessions operate on the currently loaded scripts, and new sessions are offered the new script. When the last active session on an old script set completes, the entire script set is flushed from memory, so you can operate scripted servers without downtime for rebuilding script images in memory.

DialFOX DialFOX is an express dialplan report generator that is used with the Asterisk PBX system. It is able to make an inventory of any device (such as SIP phones, softphones, and ATA) that is active in a local network. It lists their extensions, IP address, username, caller queue, device info, and comments. It can easily access with a mouse click to any SIP device that is found in the LAN. Furthermore, DialFOX provides additional information about phone devices like firmware release, key functions, and many more. DialFOX replace each sheet that are maintained by hand like hosts file and attaining this unnecessary.

Ekiga Ekiga is a SIP and H.323 compatible VoIP, IP Telephony, and VideoConferencing application that allows you to make audio and video calls to remote users with SIP or H.323 hardware and software. It supports all modern VoIP features for both SIP and H.323. Ekiga was formerly known as GnomeMeeting.

G-page g-page is a client/server application designed to send text messages to pagers or SMS (short messaging system) enabled PCS phones. It supports the SNPP, WCTP, and SMTP (email) protocols, and works on a stand-alone workstation or across a network. The home page has a list of paging providers for the USA and the protocols they support.

GNU Gatekeeper The GNU Gatekeeper is a free H.323 gatekeeper based on the OpenH323 project. You can use it to manage a Voice-over-IP network and let endpoints (e.g., Netmeeting) communicate through symbolic names. It also has an external interface for billing and other applications.

Gastify Gastify is a client for app_notify, an asterisk extension. It sits in the notification-area of the gnome-panel and displays a libnotify popup when a call arrives. By the way it logs all calls.

Gnucomm The GNUCOMM project, currently in its preliminary stages, aims to provide the standards-based free software necessary to enable the switching and transception of multimedia streams for use in telecommunications applications such as voicemail and video conferencing. The goal is not just to replace proprietary telecommunications servers and clients, but to provide better solutions to common telecommunications problems. The first goal, with version 0.1, is to establish a functional communications system. A voice respose system has already been completed; the next projects are a fax server, VOIP client, SMS server, small business scripts, and packging and documentation. Version 1.0, a functional spec for a new architecture, was completed in June 2000.

Ihu I Hear U (IHU) is a Voice over IP (VoIP) application for GNU/Linux, that creates an audio stream between two computers easily and with the minimal traffic on the network. The main features are:

• Peer-to-Peer: the communication takes place directly between the computers (UDP and TCP both supported), without need of session protocols (such as SIP or H323) or other servers in the middle.
• Good audio performance: IHU was born to give the best audio performance, low latency above all. For this purpose IHU is compatible with ALSA, now the default GNU/Linux sound architecture, but also with JACK, a low latency sound server. For the audio compression, IHU uses Speex, a codec optimized for speech (and completely free and open source).
• Crypted stream: you have also the possibility to Encrypt/Decrypt the stream using a fast hybrid cryptographic system (RSA + Blowfish)
• Command-line support: the Qt GUI is not strictly necessary, you can run also a textual IHU from command-line (for example if you need to run the program on remote computers).

The possibilities of use of IHU are endless, for example you can use it like a phone to talk with your friends all around the world, or at home/work, to talk between computers in the LAN, etc.

Isdnserver 'Isdnserver' is a server that can decode the hex values provided by your ISDN channels (B or D channels, depending on how your configuration is set up). It can send data like the phone numbers, the charging units, and the duration of a call to a predefined port, the console, or a user defined device. It can also be used as an answering machine.

Isisdial Java Isisdial Java generates the standard telephone tones via your computer's sound device. After turning on your phone to get the dial tone you simply hold it up to the computer's speaker and the phone number will be dialled for you. Written in Java with a command line interface, Isisdial Java is cross-platform and has minimal system requirements.