Search Results for "javascript open source"
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57 projects for "javascript open source" with 2 filters applied:
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ictagent
A WebRTC-Enabled Chrome Extension for Seamless Call Center
ICTAgent is a advance webphone offered as chrome extesion and when installed it will convert all available contacts as click to dial. This extension will embed WebRTC based VoIP Phone in browser which will remain connected to the main server, and will be responsible for inbound and outbound calls, transfer calls, sending DTMF as well sending Fax. It can harvest contacts from the random pages. Agent can access contents, contacts from a single application. Browser can automatically fetch... -
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Open Phone Abstraction Library (OPAL) is a C++ multi-platform, multi-protocol library for Fax, Video & Voice over IP and other networks. Also included is the Portable Tool Library (PTLib) which is a C++ multi-platform abstraction library and collection o
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Elastix
Unified Communications Server
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. -
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jfBroadcast
VoIP/SIP Autodialer
VoIP/SIP AutoDialer. Broadcasts a message with the option to transfer person to another number. Now includes new survey options. -
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FreePBX (formerly Asterisk Management Portal) is a project to bring together best-of-breed applications to produce a standardized implementation of Asterisk complete with web-based administrative interface.
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PolycomVVXControl
A command line utility for remote controlling Polycom VVX IP phones
Application to remote control Polycom VVX IP phones via their web interface (using HTTPS). This application is initially intended to perform certain actions on phones running in Microsoft Lync mode. These actions include: * get device information * get status * sign in using PIN authentication * sign out * reboot * factory reset It also supports performing actions in batch reading data from a CSV file. -
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Asterisk CDR import
Simple web interface for mapping CDR from VoIP provider to Asterisk
This is a simple tool to map information for the cost of phone calls given by your VoIP provider, to the call log inside your Asterisk. As a result, you will know how much each of your employees has spent for phone calls. Additionally, this tool shows information about the assigned DID and CLIP numbers in Asterisk. This is helpful to identify obsolete, duplicate and multiple assignments. -
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MumPI - Mumble PHP Interface
An administration web interface for Mumble VoIP servers in PHP
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… -
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jpbxlite
Java VoIP/SIP PBX system (replaced by jfPBX)
jPBXLite is a VoIP/SIP PBX. Supports SIP extensions, voicemail, trunks, conferences, queues (ACD) and an IVR system. Support video conferencing with jPhoneLite/1.4.0. NOTE:THIS PROJECT WAS RENAMED AND IS NOW jfPBX. Please go to jfpbx.sourceforge.net -
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Twisted Protocols for communication with FreeSWITCH PySWITCH is a Twisted Python library that allows you to communicate with FreeSWITCH using inbound and outbound EventSocket connections.
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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.
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PHP 2 Way Webcam Video Chat
1 on 1 Webcam Videochat Script with P2P Support
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. -
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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
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baresip
Baresip is a modular SIP User-Agent with audio and video support
Baresip is a portable and modular SIP User-Agent with audio and video support. the latest source code can be found here: https://github.com/alfredh/baresip -
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Enhanced Callback
Adds enhancements to Freepbx callback script
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ECharts is a state machine-based programming language for event-driven systems derived from the standardized UML Statecharts language. ECharts has a proven track-record in a large-scale commercial deployment. Take a look at what ECharts has to offer!
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Eolos VoipAudit
Universal Framework to audit Voip protocols's security
Visual Framework Tool to scan/sniff address space, enumerate users, crack credentials, pattern based dial spoofing and security reporting for Voip protocols. This software is intended to give a general framework to build and plug Voip protocol analizers in order to fix security issues and enhance VoiP platforms confidence. It IS NOT intended to be a cracking tool for malicious system breakers, but a really software MUST for security people to assure Voip platform security. -
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cdr2serial
Send Asterisk CDR to serial device
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. -
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SMPTE 2022-1 library for VLC & co
Optimized and cross platform SMPTE 2022 FEC library in C, Python, Java
Project moved to GitHub. https://github.com/davidfischer-ch/smpte2022lib -
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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
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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.
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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
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Cisco IP Phone Inventory Tool
Cisco IP Phone Inventory - Serial Number, MAC, Model, and more...
Project moved to github: https://github.com/vloschiavo/cipit -
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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.
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VOIP client/server in python >= 2.6. Audio in/out: ossaudiodev (UNIX like) or SoX. Network: bzip2 compression, speex or ogg audio compression, you can configure all, minimum bytes per second: 350-400 in speex mode: U8, 6 kHz, quality 0, bzip2, buf 4K
