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Printout selection
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Parameters [] 
limit:

The maximum number of results to return
offset:

The offset of the first result
link:

Show values as links
headers:

Display the headers/property names
mainlabel:

The label to give to the main page name
intro:

The text to display before the query results, if there are any
outro:

The text to display after the query results, if there are any
searchlabel:

Text for continuing the search
default:

The text to display if there are no query results
import-annotation:

Additional annotated data are to be copied during the parsing of a subject
propsep:

The separator between the properties of a result entry
valuesep:

The separator between the values for a property of a result
template:

The name of a template with which to display the printouts
named args:

Name the arguments passed to the template
userparam:

A value passed into each template call, if a template is used
class:

An additional CSS class to set for the list
introtemplate:

The name of a template to display before the query results, if there are any
outrotemplate:

The name of a template to display after the query results, if there are any
sep:

The separator between results
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ADM false
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.
ATSLog false
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.
Ant-phone false
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.
Asterisk false
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 false
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 false
* 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 false
'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 true
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.
Ccaudio true
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 true
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 true
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 false
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 false
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 false
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 false
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 false
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 false
Gajim is a Jabber/XMPP client written in PyGTK. Gajim works nicely with GNOME, but does not require it to run.
Gastify false
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 true
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 false
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).
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 false
'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 false
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 false
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 false
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?

Community contact

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 false
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 false
'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 false
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 false
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) false
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 false
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 false
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 false
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 false
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 false
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 false
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 true
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 false
A complete SIP and media stack written in C, and mainly targeted for small footprint/embedded developments.
Phpagi false
' 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 false
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 false
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 false
* " 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. "
SER or SIP Express Router is a very fast and flexible SIP (RFC3621) server. It handles thousands of calls per second on low-budget hadware. A C shell-like scripting language provides full control over the server's behaviour. Its modular architecture lets users load only the required functionality. The following modules are available: digest authentication, CPL scripts, instant messaging, MySQL support, a presence agent, Radius authentication, record routing, an SMS gateway, a Jabber gateway, a transaction module, a registrar, and user location.
Sendpage false
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 true
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 false
SmsSend allows you to send free SMS to any GSM, connecting to Internet sites using scripts.
Sofia-sip false
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 false
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 false
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 false
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 false
'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 false
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.

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