Search Results for "web sip"

Lightweight sip proxy, location server, and registrar that provides a reliable and scalable SIP infrastructure for telephony carriers, communication service providers, and integrators. Routr provides all the tools required to deploy your VoIP network, including a command-line for remote server control. It can also be controlled via a RESTful API or a web-based GUI. Routr includes all tools for deploying your VoIP network. It offers remote server control via command-line, RESTful API, or a user...

*Until you install the operating system, the default user = live / no password. DragonOS FocalX (22.04) and DragonOS Focal (20.04) are out-of-the-box Lubuntu based x86_64 operating systems for anyone interested in software defined radios. All source installed software is located in the /usr/src directory while the remaining software was installed by package managers. What is DragonOS and why do you want it? The shortest distance between two points is a straight line. DragonOS...

A C++ framework utilizing Design Patterns for creating Linux and Windows communications applications that contain Dialogic® products. Includes media and network classes (analog, digital, SIP, H323), multithreaded event handling, distributed app support.

Elastix is a software-based PBX powered by 3CX and based on Debian. An open-standards solution, Elastix is an easy to install and manage UC system compatible with popular IP phones, gateways and SIP trunks. Elastix is complete with unified communications features such as integrated WebRTC video conferencing, chat, presence and softphones and smartphone clients for Windows, Mac, iOS and Android.

VitalPBX is a free telephone and communications PBX system for companies. It is a complete platform that can be installed on the physical hardware on the site or as a hosted application. VitalPBX acts as the upper layer interface for the Linux base and then Asterisk (one of the most popular communication toolkits in the world). For this reason, VitalPBX is the graphic user interface between you and the complex world of modern communications. VitalPBX will help you implement a secure...

Vodia PBX has included a complete suite of business telephony features for on-premise and cloud-based telephony deployments. Sign up at https://vodia.com/en/register and receive your free Vodia license.

Accent (Advanced Component Control Enhancing Network Technologies) is a comprehensive tool suite that interfaces a variety of communications systems and allows these systems to be controlled through goals (high-level user aims) and policies (lower-level system rules). Accent has been applied to the domains of: o Call Control: for telecommunications, particularly call control in Internet telephony o Home Care: for home automation and telecare, particularly for domestic appliances...

Enterprise telephony recording and retrieval system with web based user interface. The project currently supports recording voice from VoIP SIP, Cisco Skinny (aka SCCP), raw RTP and audio sound device and runs on multiple operating systems and database systems. It can record audio from most PBX and telephony systems such as BroadWorks, Metaswitch, Asterisk, FreeSwitch, OpenSIPS, Avaya, Nortel, Mitel, Siemens, Cisco Call Manager, Cosmocom, NEC, etc... It is amongst others being used in Call...

Andspoilt is a command line user interface designed to easily exploit android devices. Run interactive android exploits in linux by giving the users easy interface to exploit android devices Uses an intergration with Metaspoilt Framework by giving the user an easy interface to create payloads and launch android exploits. Current Additional feature is a simple web server for file distribution.

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.

... someone needs a SIP video phone. It has been used to make calls as an ActiveX plugin, but it is pretty easy to put inside a Mozilla/Chrome plugin to make browser-based calls from a web page as well. You can drag-drop it as a control into any Windows program in Visual Studio. Do what you want with it, no need to bother me about it.

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

In this project a real time VoIP signaling analyzer and alarming system was developed which deals with SIP signaling. . The popular C library WinPcap was used to build up the program. The packets were stored in MySQL database through MySQL shell.

Fluent is a suite of tools for the management of diverse VoIP networks. Fluent RTE, a reporting and templating engine, is at the core. Add-on modules are available to manage provisioning, CDR, feature sets, routing and more.

Dubtel One is a Cloud API for quickly building advanced voice and chat applications. The API was developed using industry standards including XML, SIP and XMPP.

... 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.

CPLed is an OpenSIPS tool for editing CPL scripts in a friendly and easy graphical way. It can be used as a standalone application or embedded in a web page as applet. It also provide CPL script transport functionalities via SIP and HTTP protocols.

Kimono is an application that aim to give user peer-to-peer services by a transparent web based interface, and that let developer writing new peer to peer services as an extension of the application.Kimon0 use Mozilla Framework and a SIP DHT library.

This project is an implementation of the Sip Back-to-Back User Agent. The B2BUA acts as a user agent to both ends of a Session Initiation Protocol (SIP) call.

Web interface for administrating OpenSER SIP Server.

SIP (Sample Inventory Program) is a web application for managing mass quantities of lab samples. It is the first out-of-the-box Zope 3 app, and is field-proven. SIP may be used as-is, or as inspiration for other Zope 3 apps. (e.g. O/R RDBMS storage)

The purpose of this project is to write a plugin for the Openexchange (OX) groupware server (http://www.openexchange.com) that supports various VoIP (SIP) functionality, like direct dialing a SIP telephone, broadcasting, messaging, etc.

This library aims to implement RFC 2543 (SIP) and some of its extensions using Java

Tools and libraries for dialing URI addresses directly from telephone's numeric keypad w/o phone directories. Encoded uri (for instance, SIP VoIP phone number) consists of decimal numbers only and can be entered from any telephone or other device.

Current vTiger CRM integration is to control the Asterisk (soft) PBX to establish and log calls. This call is established by the vTiger user of the vTiger web based GUI. FreeSWITCH can be configured as a similar soft PBX and so the goal is to get vTiger to cause Freeswitch to establish SIP connections from the vTiger user's phone to the customer phone.