Open Source BSD VoIP Software - Page 5

openssl is well-known allover the world,but it is huge,complex.our target is develop a tinytls,it is small,simple,good understanding,and usually for embedded system.But it can support most openssl features.You can add some other features easily.

Turn.Message is .NET parser / generator of the TURN (Traversal Using Relay NAT) messages written in C#. Compatible with [MS-TURN] extension and clients based on Microsoft UCCAPI SDK.

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.

A Python library to create sophisticated multilingual IVR applications. NOTICE. The repository is frozen, please find the latest version of the software at https://github.com/sippy/vapp

This is an initiative to assemble/develop the cheapest, easiest to setup, easiest to manage, scalable, Open Source, standards-based, wireless local do-it-yourself telephone company toolkit in the world.

A bash shell script which will read active call destinations in E.164 format from a source (such as Asterisk), then run the called number through some plain text and MySQL databases to find the best latitude/longitude match. It then creates a .html file with Google Maps API JavaScript code which will display a single call origination point and the active destinations (calls) on a Google Maps world map.

VoIP Honey project provides a set of tools for building an entire honeynet, thus includes honeywall and honeypot emulating VoIP environments such as Asterisk PBX or OpenSer with fully configurable connections. Voip Honey runs on GNU/Linux and Windows Systems. It can be compiled for Mac OSX as well.

An XCAP server is used by XCAP clients to store data like buddy lists and presence policy in combination with a SIP Presence server that supports PUBLISH, SUBSCRIBE and NOTIFY methods to provide a complete SIP SIMPLE server solution.

YaaCs, developed by Demetra www.opinioni.net, survey research via phone CATI (also Voip) and web CAWI and for telemarketing. Develop questionnaire for survey, create probabilistic sample, monitoring Interviewer, export survey data in statistical.. It's integrate with www.modsurvey.org. Sviluppato da Demetra www.opinioni.net per indagini CATI/ CAWI e CAMI. Vi permette di programmare dei questionari elettronici, campionamenti, monitorare gli intervistatori, esportare dati in formati statistici. E' integrato con www.modsurvey.org.

The Agispeedy is an AGI Application Server implemention in asterisk and written by Pure PHP. it inconceivable faster than normal agi about 1000%, and improves performance.

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.

astercc is a realtime billing solution for asterisk, could work with all kinds of asterisk based system and no need do any change to your original system. astercc could be used for hosted callshop solution, pbx billing solution.

A simple lightweight instant messaging and VoIP client wirtten in Java. It implements the SIP protocol as described in RFC 3621 and other related RFCs and implementation requierements.

Webface witch extract CDR (Call Detail Records ) directly from MySQL base. May be used as SOHO billing system.

cdr2serial sends Asterisk CDR serial port for the use of software for hotels. They have been used libraries asterisk-java for agi events and jssc for send via serial port.

############################################### #--- CYBER-WRT by Christian Zeler / Comdif Telecom ---# #----------------- http://cyberhotel.comdif.com ------------------# ############################################## You can find 2015 news Cyber-hotel versions on http://cyberhotel.comdif.com including cyber-hotel on freepbx - Cyber-hotel decktop. . Cyber-wrt is a complet Asterisk IPBX and Hotspot system embedded in DD-Wrt router. Last version is build for ASUS RT-AC56U router It is working on Debian with asterisk pbx and original captive portal system to offer a reliable Hotspot system. It's first an original IPBX GUI with all features needed for company PBX. it include if you need a Hotel GUI offering what a Hotel PBX must do billing, check-in check-out, wake-up, housekeeping, direct call in room and many more. It include also a captive portal system for internet access controle using one lan port on your router. Try it, you will not belie

Allows OnLine monitoring of all kind Asterisk functioality: - Real-Time channel monitoring; - Queue Agents control; - Conference Administration; - Config edition; - Asterisk Core monotoring;

VoIP/SIP AutoDialer. Broadcasts a message with the option to transfer person to another number. Now includes new survey options.

Java SIP stack, originally based on mjsip.

This is my test just for testing ....

Asterisk Gateway Interface often used when it is necessary to build complex IVR system. Nanoagi will help to do it.

node (nodejs) non-blocking RADIUS client wrapper

RUMPI: Real Time Multi Peer Interaction. RUMPI want to be as ubiquitous as Flash-browser-plugin, but enabling browsers for real time communication between peers: VoIP (IAX2) and Real-time video Streaming. RUMPI is not a plattform, is a Browser plugin,

This C++ library has been designed as a Chrome SIP stack. Sippet is an open-source SIP User-Agent library, compliant with the IETF RFC 3261 specification. It can be used as a building block for SIP client software for uses such as VoIP, IM, and many other real-time and P2P communication services. The main target was to enable Javascript applications to use UDP, TCP and TLS transports along WebSocket. Existing SIP solutions for the browser are forced to use the WebSockets API to send/receive SIP messages. But for many service providers, the WebSocket protocol could not be a feasible solution due to scalability, support and other business constraints. By the way, the stack can be used as a full client-side C++ Stack (full-featured) without Javascript integration.

Git repo: https://github.com/asipto/siremis Web management interface for Kamailio (OpenSER) - handle subscriber profiles, access control lists, accounting, least cost routing and load balancing, monitoring charts, xmlrpc communication with SIP server