Open Source Linux VoIP Software - Page 6

C++ classes for control multiply SIP sessions, using eXosip library. Provides interface for creation SIP session, event processing

The Mumble PHP Interface (MumPI for short) allows you to manage your Mumble servers via a webinterface. Users can register, upload textures, and retrieve account data. Admins can manage virtual servers, registrations, admin accounts, online users…

MurMan, the Murmur Manager, is a user friendly, robust, and attractive web management interface for Murmur servers based on PHP, using Murmur's (server component for the Open-Source voice communication Mumble) Ice RCP.

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

Connecting NMS boards to the asterisk PBX via the Woomera Protocol

DAHDI Linux Drivers for Rhino Telephony Cards

Nhabla-Queue: Is a Asterisk report system for inbounds call center campaigns, provide many reports with useful information and graphics for analysis, which can be printed on PDF.

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

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.

The Open Source Contact Center Solution (OMniLeads) )https://www.omnileads.net/

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 UC Framework (openucf) is a unified communication server which supports telephony, presence, instant messaging and contact management.

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

This project is to make a open source WiFI phone software. It includes open source VoIP signaling module,P2PSIP module, GUI, Wifi module and all WiFi phone related intersting functions. Welcome to join us!

OpenAL mixer aims to provide a multiplatform API built on top of OpenAL to get/set volume values (including support for capture devices) and list, select and mute input and output lines.

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.

Pronounced "Open Spippy" (like "skippy" but with a "p" instead of a "k"): A reference implementation of the Session Peering Provisioning Protocol binding for SOAP being developed in the IETF's DRINKS Working Group.

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

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.

A web-based system that allows you to create groups of exten and generate reports of telephony usage, simple and easy to implement.