Search Results for "gnu linux" - Page 2

BI4sftp enables any Asterisk PBX to execute a scheduled secure file transfer of the Call Detailed Records related to the telephone traffic. These data can be collected and analyzed by the BI4Data Service available through web interface (www.bi4data.com)

skiputler makes your Topcom "Butler 4012 USB VoIP" phone work on Linux and Mac OS X. It's a userspace driver which connects the phone to Skype.

tui is a user interface for mobile phones with keypads. It runs on text console or under nano-X, and communicates with modem part with AT commands. It runs well on Siemens SX1 running linux, and allows you to call from it and answer calls.

ss7mon will implement the following features for SIGTRAN and SS7 family of protocols: - Message sniffer. - Call flow tracing. - Traces DB. - Distributed tracing. - QoS monitoring. - KPI monitoring. - Web interface. - CLI command line interface. - ...etc.

SIGTRAN based Short Message Center (SMSC) - proof of concept This is home of my thesis project which had goal of doing open source based Short Message Service Center to be connected into real mobile network (2G, 3G, ...). As being developed in very limited schedule (bachelor thesis work) it was limited to have very basic functionality: receiving message and deliver it immediately as-is to the receiver. There is no message caching nor different message charset support. It uses...

The 7-Scales project aims to establish a simple to use and easy to extend base for development, testing and executing of SS7 scenarios. It is conceived as a kind of pool of active knowledge in the area of telecommunication processes that can be modeled by MAP, INAP and CAP. This Project supports two modes of operation – simulation mode to develop and tests scenarios, and network mode to execute scenarios against the network. Communication with the network use a light version of SCTP and...

Project moved to GitHub. https://github.com/davidfischer-ch/smpte2022lib

KAMAILIO (OpenSER) - robust, secure and scalable Open Source (GPL) SIP (RFC3261) server implementation with large features set (over 90 extension modules). As of May 2009, source code is hosted by GIT repository at http://sip-router.org

mmsdec is an application developed in C which can be uses to decode MMS message (Multimedia Messaging Service). With this application it is possible to extract text message, photos and videos from a raw mms message.

LuaSofia is a Lua binding of Sofia-Sip library. Sofia-SIP is an open-source SIP User-Agent library, compliant with the IETF RFC3261 specification. We decided to use git, source code can be get at: git://github.com/ppizarro/luasofia.git

Libre Software to manage audio/video connection from a PC to a libre hardware real time stethoscope hardware, developed to allow doctors make a remote diagnose in real time in isolated rural areas of developing countries or for medicine teaching

sipsak is a command line tool which can send simple requests to a SIP server. It can run additional tests on a SIP server which are usefull for admins and developers of SIP enviroments. Note: the project is maintained on GitHub https://github.com/nils-ohlmeier/sipsak

An extension to the standard Asterisk PBX expanding it's capabilities to include messaging.

OSP Module enables SER (SIP Express Router) to use the Open Settlement Protocol for secure Authorization, Routing and Accounting. OSP is an ETSI protocol, Technical Specification 101 321

dtmfbox is a small softswitch application (SIP/CAPI), that can be used to control different tasks over telephone keyboard via DTMF. Mostly, it was made to run on the AVM FRITZ!Box 7170 (mipsel) but works under Unix/Linux and Windows, too.

GSM Multiplexer Daemon is intended to be used for GSM Modens. It implements (a subset of) the GSM 07.10 standard, it can be used to multiplex one or more logical channels over one physical serial channel to the modem.

Asterisk based SIP b2bua realization. It's patched asterisk and some AGI script. Features: full vovida's b2bua radius emulation, extended radius attributes, radius failover, LCR, Call failover, Codec based routing and other useful things.

Very fast and very flexible SIP server

PartiSIPation is a sip user agent which has an easy replaceable gui. So it can be used as a softphone as well as it can be integrated in any kind of application, e.g. online games.

Siemens EWSD PSTN Exchange can dump its configuration into regenfiles (or commandfile, cmdfile). rg2xml interprets these files and transforms information to XML

Solid PBX is a multi-platform Soft Switch Class 5 software targeted for home/corporate/operator level markets.

The ATSlog software provides a handy web-oriented interface for collecting, viewing and analysing calls for various types of PBX (Private Branch eXchange) models.

This project is for people who want to access one or more HFC-S PCI A based ISDN BRI cards over the DAHDI interface and maybe want to use other DAHDI hardware in parallel, too. DAHDI driven hardware can be used by Asterisk, FreeSWITCH, Yate and other telephony software. The HFC-S PCI A based cards are very cheap and thus ideal for private BRI users. The OSLEC is being developed since a long time ago and said to be very good. The superiority of "hardware echo cancellers" is a myth. This...

Video Applications for Asterisk (libh43m, app_h324m, app_mp4, app_rtsp, app_transcode, app_swf) H324m stack, mp4 record and playblack, rtsp support, video transcoding and swf file playback.