Search Results for "open source assembly software source code" - Page 2

asterCRM is a call center software for asterisk based VoIP system, also it has some CRM functions. It provide useful features such as pop-up, predictive dialer, click to call, extension status .... astercrm could work with all asterisk based system.

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

Session Initialtion Protocol (SIP) is widely used for telephone services over the Internet. CafeSip provides a suite of open-source tools and applications for creating customized SIP services and applications using the Java.

This project serves to provide respins of some common open source PBX projects to include additional drivers, functionality, and software not found in the original projects.

OpenCallshop for VoIP Switch was started to be an alternitive to VoipSwitch Inc. closed source callshop systems, You can extend development of this project and customize it to fit your needs. if you would like to get a custom developed callshop desktop or web software do not hesitate to contact me. if you're reseller this product consider a donation.

SRPP: A library to implement SRPP (Secure Real-time Padding protocol) support. It is aimed at providing application-level packet padding for VoIP traffic, so as to thwart traffic analysis attacks on VoIP calls.

A JNI wrapper for pjsip. You can use this wrapper to develop Java applications using the pjsip library. At the moment only the pjsua API is implemented. If you would like to obtain a commercial license, or need customisations, please contact us.

Java implementation of MIKEY (Multimedia Internet KEYing, RFC 3830).

[es ] Aplicación web central para particulares. [en ] Core web application for individuals. [fr] Application Web centrale pour les personnes. [pt] Central de aplicação web para indivíduos. [ar] المركزي تطبيق ويب للأفراد

Open source software components of the AastraLink Pro 160 Supplied from CDROM "71-001049-02 REV01" by Aastra Telecom

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.

Castadiva is a test bed based on low-cost off-the-shelf devices, which is used to test protocols developed for MANETs. Castadiva is completely compatible with the file format used by the ns-2 simulator.

It is an appliance software that integrates the best tools available for Asterisk-based PBXs into a single, easy-to-use interface. include astercrm astercc voicemail,fax,et

The Java Teamspeak 3 Server Query Software Development Kit is a Java Library that provides Classes to access the Teamspeak Server Query for easy integreating TSSQ into Java Desktop- or Web-Applications.

OaxRom WebSMS is a software that allow send multiple SMS from a web page, you will only need capture category, link with contacts and then you can send sms messages from a webpage selecting a category then will send sms to all contacts of the categor

...The OpenSIPS control panel fulfill this need and it is not wise to maintain two GUI projects. Our team will support the opensips-cp initiative to avoid splitting the community between two open source projects for OpenSIPS GUI. I would like to say thanks to all the people that in some way contributed with the project. The svn code is available to anyone who wants to maintain the project. Most of our development efforts are now focused in a new commercial product based on OpenSIPS called SIP Pulse, completely rewritten in Java and Glassfish www.sippulse.com supporting pre and post paid users with billing and reseller portal.

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

Eclipse plugin providing form editing and validations on JAIN SLEE 1.1 deployment descriptors. This plugin requires Eclipse 3.5 (Galileo) or later.

The Asterisk Config PHP-Parser claims to be a simple but effective function writen in PHP non-OOP that is capable to parse any standard .conf Asterisk configuration file an drop the data into a multi-dimensional array.

Twinkle is a VOIP based telephony software on linux. This project will install this software on MAC OS X using fink package management system. This will be included in stable tree of fink.

Vorilon is a Voice Teleconferencing software consisting of a Client and Server applications to be used for Game Voice chatting and General Voice Teleconferencing.

A command line SIP/H323 softphone capable of sending and receiving audio files as well as sending out of band DTMF digits. Supports a BNF format configuration language for scripting call scenarios. Useful for example system testing.

This project aims to build a software reference implementation of the EBU standard for the transmission of high quality, low latency, audio streams over IP networks. (EBU-tech 3326)

FATS - FATS is a Twisted and Fast Asterisk's Telephony Services. Project contains implementation of FastAGI, AMI protocols for the Twisted framework. Using it you can develop fast and pretty services for the Asterisk IP-PBX.