- '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 (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' 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 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 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 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.
- 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.
- 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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