Search Results for "c-sharp" - Page 3
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Showing 411 open source projects for "c-sharp"
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AVAYA CMS API Server
CMS API Server with web site
Free version of AVREMO.CMS.API exposing REST Services. With this software you can browse for all CMS reports. Design a report with AVAYA CMS Supervisor designer, application automatically expose data as XML or JSON. You can also browse reports with your preferred browser (desktop or mobile) and execute CMS operations like modify dictionaries, modify agent skills, modify VDN/Vector associations and more. Application package include : - A responsive WEB server interface (compatible... -
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integral
Fast CGI-based AsteriskPBX cluster administration tool.
Allows OnLine monitoring of all kind Asterisk functioality: - Real-Time channel monitoring; - Queue Agents control; - Conference Administration; - Config edition; - Asterisk Core monotoring; -
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usb-b2k
programm to handle a USB-B2K box
The program consist of an API (daemon) to handle a VOIP box of the type USB-B2K which allows to plug a classical telephone on an USB port (skypebox, wangobox, yealink...) in order to integrate this with any type of program of the type VOIP under GNU/LINUX. The functionalities are the following: 1. Switching to mode USB/PTSN 2. Detecting Pick up/ Hang up signal from the telephone 3. Generating Pick up tune 4. Generating Ring tone. 5. Handling telephone dial. This project is... -
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oSKY
Framework implementing SS7 over IP & SIGTRAN interworking and trunking
oSKY (OpenSKY) is a full communication framework to implement PSoTN (Public Social Telephony Network). In contrast with PSTN (Public Switched Telephony Network), oSKY intends to implement an open paradigm in the internet domain, gaining more than 100 years of experience from traditional telephone network. oSKY goals: -enhanced security (SSL, SCTP) -embedded NAT tunneling -Global title discovering on Internet style -Peer2Peer mode and Gateway mode -more... -
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mod_MumbleLink
Positional Audio Communication for Minecraft with Mumble
A Mod so that Minecraft now natively supports Mumble's positional audio feature. This means: Directional and positionally attenuated VOIP in relation to the game world. Please visit the Forum for information about the newest Version! Main Forum-Thread: http://www.minecraftforum.net/forums/mapping-and-modding/minecraft-mods/1272675 TheSkorm's Fork on GitHub: https://github.com/TheSkorm/mod_mumblelink Mumble: http://mumble.sourceforge.net Donations are greatly appreciated:... -
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GNU Telephony
GNU Telephony Services and Applications
GNU Telephony intends anyone to use free as in freedom software for telephony, and to do real-time collaboration freely and privately over the Internet, with the freedom to do so on any platform they choose to use. -
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Airean
GSM air interface
Airean is a proof-of-concept application showing that eavesdropping of GSM traffic (voice calls, mobile locations, SMS messages) is possible using a standard desktop PC with a small investment in hardware. The application requires a hardware interface to capture signals. Currently the following devices are supported: * Ettus Research USRP * Great Scott Gadgets HackRF (will be implemented when the device is released) Airean has the following capabilities: * Scan the GSM spectrum *... -
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Voxy - the simplest way to integrate VoiceXML applications in Asterisk. Configure your dial plan with the URL of your VoiceXML application and it's done.
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Your monitoring isn't a stack. It's a pile. Fix that.Replace Datadog, New Relic, and Sentry without adding three more dashboards.
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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)
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baresip
Baresip is a modular SIP User-Agent with audio and video support
Baresip is a portable and modular SIP User-Agent with audio and video support. the latest source code can be found here: https://github.com/alfredh/baresip -
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Project moved: Find the latest code at https://www.h323plus.org/ Follows on from now depreciated OpenH323 and designed as a drop in replacement for OpenH323. Development of advanced open source H.323 including application sharing, video conferencing and incorporates new research and development work. Project moved: Find the latest code at https://www.h323plus.org/
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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.
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DTMF detector and generator. Was used standard C++,so DTMF detector and generator completely cross platform.It used robust and reliable fixed-point algorithm, so can fast work on fixed-point architecture.
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Sipp is a performance testing tool for the SIP protocol. Its main features are basic SIPStone scenarios, TCP/UDP transport, customizable (XML-based) scenarios, dynamic adjustment of call-rate and a comprehensive set of real-time statistics.
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One of the first secure telephony programs. Not limited to VOIP but can also directly encrypt a telephone line by using modems. Calls are encrypted using Diffie-Hellmann key exchange and several symmetric ciphers. Download is possible from Germany only, see homepage
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Teamspeak 3 Love Plugin
A plugin for Teamspeak3. This plugin allows you to autofollow a user.
A plugin for Teamspeak3. This plugin allows you to follow a user while he switches through channels. For the love menu just right click any name in the server view. -
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This application use sipp. The GUI base application try to create xml and csv files easily and start scenario which are selected. sipp and mono have to be installed on your PC. If you want to send RTP packets, you should copy pcap files to same folder where running sipp_gui.
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SIP Anonymization Tool (SiAnTo)
Small and effective program for SIP traces anonymization
The Session Initiation Protocol (SIP) is a signaling communications protocol widely used nowadays for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks. A good way to design optimization techniques for SIP deployment would be to analyze SIP traffic from existing networks. However, publicly available analyses of SIP traffic are rare and thus not a lot of knowledge exists about typical behavior of a SIP server (as opposed to, for... -
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QtMoko is distribution and software stack for Linux based SmartPhones
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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.
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Seagull is a multi-protocol traffic generator. Especially targeted towards IMS, Seagull supports Diameter (RFC3588 and all applications) over TCP/SCTP and IPv4/IPv6 , TCAP (over SS7 or Sigtran), XCAP over HTTP and Radius.
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ss7mon
ss7mon an SS7/SIGTRAN signalling monitor
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. -
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sigtran-smsc
PoC: Short Message Service Center (SMSC) over SIGTRAN
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...
