Open Source BSD VoIP Software - Page 3

Fideliphone: High Quality Internet Telephony

If you want to quickly set up a multi-tenant PBX on matter of few minutes, this is a suitable choice. - Demo environment. You can start to add extensions, gateways and dialplans, and quickly have a fully working PBX environment. - Import previous FusionPBX backup to Live environment. After this you will have a quickly working PBX on few minutes. - Install on hard disk. With this option you can install the environment to Hard Disk. Username and password for PostgreSQL is fusionpbx / fusionpbx . Type when the wizard asks for these credentials. root password is 'password'

Google Contacts to Grandstream xml Phonebook Format Converter

Does not work because there is no master server! A P2P Messenger which uses JXTA. It provides secure RSA/AES end-to-end encryption and VoIP-Capabilities.

This is a script to engage telemarketers and waste their time. Also records the conversation for listening later.

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.

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

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.

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

Lwazi is a robust telephony platform aiming to facilitate speedy development of experimental applications without sacrificing power by combining Asterisk with the MobilIVR Python interface bundled into a single build with a unified control interface.

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

This project aim to provide a web based application to plan Conference Call using Asterisk application MeetMe(). The engine is based on Vaadin language to improve the portability of the application, Hibernate and some AGI script based on PHP.

Mobicents is the leading Open Source VoIP Platform. It is the First and Only Open Source Certified implementation of JSLEE 1.1 (JSR 240), and SIP Servlets 1.1 (JSR 289). Mobicents also includes a powerful and extensible Media Server.

Global mesh development software and hardware platform from developing global wireless mesh network using multiple frequencies.

A modularly designed system providing command/trigger functionality and response mechanism via SMS text messages

MumurWeb will be a comprehensive webinterface for Murmur servers based on PHP. Murmur is the server component for the Open-Source voice communication Mumble.

Olyo is an open source project which aims at implementing a prototype for Peer-to-Peer Session Initiation Protocol (P2PSIP) and providing real-time multimedia services over Internet based on this prototype. Olyo is promoted by MINE lab, BUPT.

This is a repository for QueueMetrics add-ons that are contribuited by the community and are open-source.

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

Open Unified Recording (OUR) is a full featured Linux based Open Source VoIP/SIP Call Recording engine, indexing and retrieval system. The system resides on the network and passively captures SIP sessions. Project is sponsored by UnifiedRecording.com

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.

The goal of the project is to create a high-performance, open-source and standards-compliant implementation of a Home-Subscriber-Server (HSS) for use in a IMS context.

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

A Web Control Panel Application for the OpenSIPS, which is intended for both system and user provisioning. It features more than 18 tools, covering important functionalities (MI,statistics) and modules (acc,siptrace,drouting,dialplan) of OpenSIPS. IMPORTANT: this is no longer the main hosting for the project. This was moved on GITHUB - https://github.com/OpenSIPS/opensips-cp .

PBXlab adds value to business thanks to its valuable contribution of technological solutions. PBXlab promotes the use of free software programs that can be tailored to the real needs of its users, speed of movement between users, at a very low cost or no cost. PBXlab offers development of add-ons and improvements to programs, leading to more full, safe and adapted to customer needs programs. We offer technological independence without having to develop products from scratch. PURPOSE: This project is designed to install the latest stable version of certified-asterisk-13.1-current + FreePBX V.12-based system in Debian 8.1 Platforms and versions tested: + 686 and amd64 + Debian 8.1 Jessie + Certified Asterisk 13.1-current - LTS + Libpri 1.4.4 + DAHDI COMPLETE LINUX Current + FreePBX 12. + Avantfax 3.3.3 + FOP2 2.29.02 + Webmin 1.760 + PHPSysInfo 3.2.2 + Fail2ban + CSF Firewall