Semantic search
- 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
- 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
- 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 software).
- 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).
- 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 Desktop, formerly known as the SIP Communicator and briefly known as just Jitsi, is a VoIP and instant messaging application that supports protocols such as SIP, XMPP/Jabber, AIM/ICQ and IRC. It can handle every firewall, and has many other useful features. This package is maintained by the community. Video conferencing capabilities evolved out of this original project and are maintained by the Jitsi team under 8×8 which acquires the whole Jitsi Technology in 2018. So the said Jitsi can also refers to a set of free software projects including Jitsi-Meet and Jitsi-Videobridge. More information at https://jitsi.org/
- Jitsi-Meet
- More secure, more flexible, and completely free video conferencing.
Go ahead, video chat with the whole team. In fact, invite everyone you know. Jitsi Meet is a fully encrypted, 100% open source video conferencing solution that you can use all day, every day, for free — with no account needed.
What else can you do with Jitsi Meet?
Share your desktop, presentations, and more Invite users to a conference via a simple, custom URL Edit documents together using Etherpad Pick fun meeting URLs for every meeting
Trade messages and emojis while you video conference, with integrated chat.
- Jitsi-Videobridge
- Build massively scalable multiparty video applications Stop mixing video channels and start using Jitsi Videobridge instead. It’s a Selective Forwarding Unit (SFU) designed to run thousands of video streams from a single server — and it’s fully free software and WebRTC compatible.
- 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 software 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.
- Nextcloud
- Nextcloud is a free, decentralized and open cloud platform. It allows you to sync your files, calendars and contacts while staying in control of your data. As a platform you can extend it with many existing apps or write your own. With federated sharing, Nextcloud allows you to collaborate with people across different Nextcloud installations. Nextcloud also provides real-time communication through Spreed.me and collaborative editing through LibreOffice Online integration.
- OpenAnswer
- OpenAnswer is a fully-functional answering service platform, compatible with Asterisk and completely open source. We've created our own powerful answering service software, built to break free of the big game players in the industry charging exorbitant rates per seat. Flexible, powerful and user-friendly out of the box.
- OpenSER
- Flexible and powerful SIP (RFC3261) proxy/server enabling easy VoIP service creation. With a modular architecture, OpenSER can be customized to fit the needs of deploying carrier grade as well as residential services.<\p>
- Osip
- The GNU oSIP library is an implementation of SIP (as defined by RFC 3261). This is the oSIP library (for Omnibus SIP), which consists of a parser and a transaction manager. It gives multimedia and telecom software developers an easy and powerful interface to initiate and control Session Initiation Protocol (SIP) based sessions in their applications. SIP is described in the RFC2543. The oSIP home page includes links to various useful SIP sites. 'oSIP' is little in size and code and thus could be use to implement IP soft-phone as well as embedded SIP software. oSIP is not limited to endpoint agents, and can also be used to implement "SIP proxy". It does not intend to provide a high layer API for controlling "SIP Session" at this step. Instead, it currently provides an API for the SIP message parser, SDP message parser, and library to handle "SIP transactions" as defined by the SIP document.
- PJSIP and PJMEDIA
- A complete SIP and media stack written in C, and mainly targeted for small footprint/embedded developments.
- Phpagi
- ' phpagi' is a PHP class for writing Asterisk AGI scripts. The class encapsulates many common AGI tasks, and adds enhanced functionality for helping to develop vertical applications and utilities.
- PodMail
- PodMail brings together telephony and podcasting by integrating with Asterisk to provide a secure podcast of your voicemail. Each time you dock your iPod, your new voicemails will be synced, allonwing you to listen to your voicemail at your convenience and without using a cell phone. If configured for public access, you can update your podcast simply by making a phone call.
- QuteCom
- QuteCom uses your broadband internet connection (DSL, Cable or WiFi) to provide telephony services. Call your friends, family, and colleagues anywhere, anytime, for free. Use video and chat features, also for free. QuteCom also offers low cost international rates (prepaid or unlimited) to land lines or mobile phones, and an SMS gateway service. The QuteComphone is an easy to use program, that enables you to call anyone, anywhere, anytime, for free. Its native use of the SIP protocol makes it interoperable with most on the known VoIP technologies. It already has built-in video capabilities, and is will offer many more features in the near future such as calling ordinary phones (land lines and mobiles) at very low prices, conference calls, user communities and innovative cellphone-based services. QuteCom is formerly known as both OpenWengo and WengoPhone.
- SIP Express Router
- * " November 04, 2008 - SER joins the sip-router project "
- " First is important to clarify: from version 3.0.0 on, Kamailio and SER are identical in terms of source code. "
- Sendpage
- To send alphanumeric pages to a pager, if an email gateway is unavailable or undesirable, software is needed to control a modem which will dial a "Paging Central", and deliver the pages using an ASCII delivery system known as "TAP". Sendpage implements all aspects of this type of software, including an SNPP client, an SNPP server, a queuing engine, a modem control engine, a TAP communication system, and an email notification system.
- Sipwitch
- GNU SIP Witch is a secure peer-to-peer VoIP server. Calls can be made even behind NAT firewalls, and without needing a service provider. SIP Witch can be used on the desktop to help create bottom-up secure calling networks as a free software alternative to Skype. SIP Witch can also be used as a stand- alone SIP-based office telephone server, or to create secure VoIP networks for an existing IP-PBX such as Asterisk, FreeSWITCH, or Yate. GNU SIP Witch is also part of GNU Telephony & the GNU Telecom subsystem.
- SmsSend
- SmsSend allows you to send free SMS to any GSM, connecting to Internet sites using scripts.
- Sofia-sip
- Sofia-SIP is a SIP User-Agent library, compliant with the IETF RFC3261 specification. It can be used as a building block for SIP client software for uses such as VoIP, IM, and many other real-time and person-to-person communication services. The primary target platform for Sofia-SIP is GNU/Linux. Sofia-SIP is based on a SIP stack developed at the Nokia Research Center.
- Speak Freely
- Speak Freely uses workstation audio hardware and network to allow bidirectional conversations right over the network. As of January 15, 2004, development of this packages has been discontinued.
- Telegram
- Telegram is a multi-platform instant messaging client. It can be used on multiple devices at same time. It supports sending messages, photos, videos and files of any type (doc, zip, mp3, etc) to people who are in the phone contacts and have Telegram. The messages are encrypted using the MTProto protocol developed by Telegram. Its features include self-destructing messages & broadcast groups. This is a desktop client program only; there is currently no free software server program for Telegram.
- TelemarketingLogs
- TelemarketingLogs is a simple telemarketing application, that allows to keep track of the calls that you made to your clients or contacts. It has been developed as a HTML5 app, using AngularJS and Foundation. Features: Works offline. Export/Import your projects to JSON. Let you organize your calls in projects. Let you manage your list of contacts. User guide inside the application.
- Tsemgr
- 'tsemgr' is a gtk application to manage the Sony/Ericsson t68 mobile phone. you can read and send short messages (sms), view and edit the phonebook, upload files via irda and bluetooth and turn your phone into a remote control for your linux box.
- TuxCall
- TuxCall is a call waiting detector that hangs up the modem connection when a voice call arrives. It reads sound from modem's speaker (must be connected to audio line input) and kills pppd, allowing you to answer the phone.
- Twinkle
- Twinkle is a softphone for your voice over IP and instant messaging communcations using the SIP protocol. You can use it for direct IP phone to IP phone communication or in a network using a SIP proxy to route your calls and messages.
- Umbrella
- Umbrella aims to be an full-fledged project management suite including the following features:
- Project and task lists
- Timetracking
- Invoicing
- Bookmarks
- Stock management
- File browser
- VoiceOne
- VoiceOne is an easy-to-use, Web-based GUI for the Asterisk PBX. It gives full control over PBX settings from the admin configuration panel. You can manage extensions, VoIP trunks, users' queues and rules sets, and dynamically create an IVR.
- WikiSuite
- The most comprehensive and integrated Free / Libre / Open Source enterprise software suite. WikiSuite™ is especially suited to knowledge-centric organizations and offers most (80%+) of the data and information management features all organizations need Key features of WikiSuite are OS & Network management, Email & Calendar, Commerce, Web & Intranet, Chat & Video Conference, Security, BPM & Analytics, Files & Sync. WikiSuite is an integrated and extensible solution.
- Yate- Yet Another Telephony Engine
- "YATE" (Yet Another Telephony Engine) is a telephony project that is written in C++ and that will support scripting in various programming languages. It is a next-generation telephony engine; while currently focused on Voice over Internet Protocol (VoIP) and PSTN, its power lies in its ability to be easily extended. Voice, video, data and instant messenging can all be unified under Yate's flexible routing engine, maximizing communications efficiency and minimizing infrastructure costs for businesses.
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