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Ccscript Heckert gnu.tiny.png
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
This is a candidate for deletion: 1. No files sources found. 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 (formely known as GnomeMeeting) is a SoftPhone, Video Conferencing and Instant Messenger application over the Internet. It supports HD sound quality and video up to DVD size and quality. It is interoperable with many other standard compliant software, hardware and service providers as it uses both the major telephony standards (SIP and H.323).
Farstream
The Farstream (formerly Farsight2) project is an effort to create a framework to deal with all known audio/video conferencing protocols. On one side it offers a generic API that makes it possible to write plugins for different streaming protocols, on the other side it offers an API for clients to use those plugins. This package provides Python bindings for Farstream.
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. You can use it to manage a Voice-over-IP or videoconferencing network and let endpoints (e.g., Ekiga) communicate through symbolic names. The GNU Gatekeeper provides NAT traversal and call encryption. It also has an external interface for billing and other applications.
Gajim
Gajim is a Jabber/XMPP client written in PyGTK. Gajim works nicely with GNOME, but does not require it to run.
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 Heckert gnu.tiny.png
The GnuComm project, currently in its preliminary stages, aims to provide the standards-based 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.
Jitsi
Jitsi (previously SIP Communicator) is an audio/video and chat communicator with full end-to-end encryption that supports protocols such as SIP, XMPP/Jabber, ICQ/AIM, Windows Live, Yahoo!, GTalk/Hangouts extensions, as well as OTR, ZRTP, etc. It can handle every firewall, and has many other useful features.
Kisdnmonitor
'Kisdnmonitor' is a KDE applet for isdnserver. It monitors the B-Channels of your ISDN cards, shows each incoming/outgoing call, any calls that arrived while the user was absent/logged out, and call time, duration and cost. It provides a list of all calls with editing and find options; you can print this list in the row order and sorting order you set up. It is linked to kaddressbook (edit amd insert). It shows the email addresse(s) of the caller in the context menu (if available), and statistics to graphically view the data.
Kontalk
Kontalk is a free software, secure and distributed instant messaging driven by the community. Kontalk protocol is based on XMPP with end-to-end encryption in both server-to-server and server-to-client. Kontalks is basically for phone, but it's also available for desktop now (GNU/Linux, Windows, and macOS).
Linphone
Linphone is an Internet phone or Voice Over IP phone (VoIP). It lets you make two-party phone calls using the Internet.
Mbuni - Multimedia Messaging Server (MMSC)
Mbuni MMS gateway is a modular software system, designed to be full-featured, efficient and simple, supporting current generation two-way multimedia messaging. Feature highlights include:
  • Phone-to-phone messaging
  • Automatic content adaptation: The server modifies message content depending on the capabilities of the receiving terminal
  • Integrated Email-to-MMS and MMS-to-Email gateway
  • Support for persistent storage of messages for subscribers (MMbox).
  • Inter-MMSC message exchange (MM4 interface)
  • Support for MMS Value Added Service Providers using MM7 protocols (SOAP or EAIF).
  • Support for integration with subscriber database to enable smart handling of handsets that do not support MMS, handsets not provisioned, etc.
  • Support for flexible billing structure through billing/CDR plug-in architecture
  • Bearer (data) technology neutral: Works with GSM/CSD or GPRS.
  • The Gateway is designed and tested to conform to Open Mobile Alliance (OMA), WAP and 3rd Generation Partnership Project (3GPP) MMS standards including: WAP: 209, OMA: MMS v1.2, UAProf v1.1, 3GPP: TS 23.140
Mediastreamer
Mediastreamer is library written in C that allows you to create and run audio and video streams. It is designed for any kind of voice over IP applications. It features RTP connectivity, audio codecs (Speex, iLBC, G711, GSM), video codecs (MPEG4, H263, Theora), I/O from soundcards, wav files, webcams, echo-cancelation, conferencing, and various other utilities. It has a modular design that makes it extensible through plugins. This is the media-streaming component of linphone, a GPL SIP video phone.
Mgcp
This package is a project to write a free implementation of the Open Media Gateway Control VoIP Protocol. It is not yet functional. Preliminary code can be obtained from the URI listed in the resource tab. There has not yet been a formal release. The Media Gateway Control Protocol controls telephony gateways (which convert audio signal from phone systems to data packets from the Internet) from external call control elements called media gateway controllers or call agents.
MySIPSwitch
The SIP Switch is an experimental stateful SIP Proxy server sponsored by Blueface Ltd to allow the use of multiple supplier SIP accounts from a single SIP login. This is a free and open source application our code is on Source Forge
The SIP Switch has call management functions allowing you to hold/resume, transfer, forward calls on the fly from a web browser.


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