Search Results for "version 2"

Legacy SVN repositories for projects from David Rowe http://rowetel.com Mailing lists for Codec 2 and SM1000.

New version 1.2. This is the code to STUNTMAN - an open source STUN server and client code by john selbie. Compliant with the latest RFCs including 5389, 5769, and 5780. Also includes backwards compatibility for RFC 3489. ICE and WebRTC ready. Version 1.2 compiles on Linux, MacOS, BSD, and Solaris. Supports the STUN protocol on both UDP and TCP for both IPv4 and IPv6. Windows binaries are also provided. Additional features are in development. This is a mirror of the code on...

Intel Integrated Performace Primitives audio/video codecs plug-in for the OPAL/OpenH323 library including G.728, G.729, G.723.1, G.722.2 GSM-FR, GSM-AMR, H.261, H.263, H.264 and MPEG4 part 2.

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.

...Es una centralita VoIP Asterisk con altas prestaciones para las Empresas. Última versión disponible 2.11.19 en la web de VozBox: http://www.vozbox.es/descarga/ Internet connection required during installation. Update version: 2.11.19 1 - Customizing bash_login and bashrc 2 - Upgrade to version libpri-1.4.15 3 - Included new module-manager FOP2-1.0.3 4 - Inclusion of new packages and dependencies. 5 - Correction of errors in the installation. 6 - Upgrade to version webmin-1.710 7 - Correction of errors in the pbx-vpn script. 8- Inclusion of new script pbx-status.

This is a web based instant 1 on 1 private online video conferencing solution. It's a solution for conducting easy to setup face to face meetings without leaving your office or home. It's the easiest and most cost-effective way to meet somebody and discuss one on one, to make a video call just by providing a private room access link.

O OpenVOIP é uma solução Open Source para que empresas possam ter um padrão de comunicação VOIP.

A simple interface to work with SIP numbers of Asterisk. Add, delete, and edit SIP numbers in the MySQL database As well as viewing CDR records Простой интерфейс для работы с SIP номерами сервера Asterisk. Добавление, удаление и редактирование SIP номеров в базе MySQL А так же просмотр CDR записей

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.

SIP over IPv4 and SIP over IPv6 are different worlds, making it impossible to register IPv4-only phones on IPv6-only services, and so on. SIPproxy64 is a proxy that translates between the two worlds. It is small enough to run inside a router.

This is a open source media gateway controller. The protocol used is MEGACO/H.248. In a later version also SIP will be supported.

A $10 Analog Telephone Adapter (ATA) that allows low cost routers such as the WRT54G to connect to an analog telephone. The application is low cost VOIP, in particular for the developing world.

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.

The project "Sippie" is an additional tool for automatically converting Wireshark-Traces to XML - scenarios, which can be easily used as infile for the OpenSource SIPp test tool and traffic generator.

I'm sad to say that, unfortunately, the serMyAdmin project is being discontinued starting in September 1st, 2010. After three years of development, the project no longer makes sense. The main reason to create the serMyAdmin was the lack of a GUI for OpenSER/OpenSIPS. 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...

The system is used to control elements of a smart home remotely, with a web or a telephony interface. Automation can also be created to automatically execute actions based on events.

LibDreamSpeak is an example implementation of the TeamSpeak2 protocol (client-side) in Java. Some day it may be used as base for alternative TeamSpeak2 (mobile?) clients.

PHP / ZeroC ICE based web administration system for Murmur Servers.

OSSEG is an IMS compatible security gateway that adds firewall, VPN, IDS, IPS, and monitoring functionalities to VoIP networks.

An tray icon tool to allow automatic Dial using an XMLRPC framework. The application is based on two separate tools: - a light desktop application - a XMLRPC server witch make an PBX abstraction. (first driver is for Asterisk)

A C++ lib for the Asterisk PBX AGI interface.

Cross-platform Java Teamspeak client

This is a SIP based conference server. Allow to host multiple conference bridges and allow multiple participants in each conference.

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,

A complete contact center solution for the Asterisk PBX, capable of handling inbound and outbound campaigns. Provides an agent toolbar for CTI. For managers an admin panel controls every aspect of the call center. Stunning historical and realtime stats.