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Maintained version of old asterisk applications ported to 1.4 that for copyright reasons cannot be included into official Digium's releases and that for some reason the author is not keeping up-to-date. The project also require Zaptel instead of DAHDI.
Mission Control is a Telepathy component implementing the Account Manager and Channel Dispatcher specifications. It stores IM account settings, brings accounts online as necessary, and hands communication channels to the relevant applications.
Addons to the Django Framework for mobile clients. MoGo was originally built to handle JP specific issues, but code to handle other locales are welcome as well. Developed and maintained by MARIMORE Inc http://www.marimore.co.jp
THIS PROJECT HAS MOVED TO https://github.com/marimore/mobiledjango
Simple testing tool to generate RTP data packets and send it via netwok interface or save into pcap file. Primarily intended for use with SIPp application to test speech quality with different codecs.
OpenBTS is an implementation of the GSM air interface (Um) that allows cellular handsets to be used directly as SIP endpoints. It uses a software-defined radio to generate its air interface and uses Asterisk or yate as its network interface.
Live-cd com Servidor SIP baseado em opensips. O Servidor de Telefone Livre transformará um simples PC numa central SIP multidominio. Estamos construindo a documentacao do projeto na wiki do sistema trac.
My-WIM is a new instant messenger service, which provides chat, white board, voice and video chat, games, file share, file transfer and many more features. Active development started!
Telepathy-SofiaSIP is a SIP connection manager (protocol plugin) for the Telepathy framework (http://telepathy.freedesktop.org). It's based on the Sofia-SIP stack.
VoIP's project with Speex codec, speech detecting and crypto coding with traffic counting. All code is a pure Java, so completely cross-platforms. Traffic is about 9 MB per hour in both ends, sources: http://www.open-source-soft.narod.ru/arrow.7z
8ix Zenith CE spells Asterisk derived IP Telephony solution.When you want robust communication, without the huge cost and recurring bill that comes with the way you do business, 8ix Zenith CE addresses your every communication need.
Prototype testbed implementation of the IETF Media Server Control (MEDIACTRL) SIP Control Framework, comprehensive of both control and processing functionality (as in IMS MRF, Media Resource Function).
Kiax is a softphone (soft phone, VoIP client) with a simple and comfortable user interface for making VoIP calls to Asterisk PBX. It depends on the iaxclient library to use Asterisk's IAX2 protocol for easy call configuration and audio setup.
KPhone is a SIP UA for Linux, supporting a multitude of features. Originally developed by Billy Biggs, it was developed at Wirlab until 2005. It is now developed by a team of volunteers in this project.
AMS 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.
RTP text/t140 Library is a reference implementation for RTP Payload Type for Text Conversation (RFC 4103). The library has sourcecode for encoding and decoding RFC 4103 data, and may be used either as a plug-in to JMF or in a separate RTP sender/receive
This is a general-purpose library for communicating with the Skype API. It is simple, cross-platform, and easy to use. The library currently works on Windows (Win32) and Linux (X11/D-BUS).
JSR180 is the standard API spec by jcp.org which is been implemented over Resiprocate Stack. Our purpose is to enable "Open SourceCode with Open APIs". By implementing JSR180 we could easily access the SIP Stack by the API documented in the JSR180 Spe