Search Results for "information"
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Showing 46 open source projects for "information"
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Ship Agents FasterGemini Enterprise Agent Platform lets you rapidly build, scale, govern and optimize production-ready agents grounded in your organization's data. The platform enables developers to build custom or pre-built agents for virtually any use case. New customers get $300 in free credits.
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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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VitalPBX
Unified Communications System - PBX System
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... -
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Stuntman - STUN server and client
High performance, production quality STUN server and client library
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... -
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MCU Media Server
SIP Video Multiconference Media Server with WebRTC support.
REPOSITORY MOVED TO GITHUB!! https://github.com/medooze/media-server Video Multiconference Media Server with WebRTC support. Provide Multiconference and video broadcasting services to any SIP service. Supports VP8, H264, MP4V-ES, H263 and H263P, continuous presence, RTMP flash broadcasting, adhoc conferences, load balancing and administrative WEB interface. JSR309 driver implementation under development. . -
$300 Free Credits for Your Google Cloud ProjectsLaunch your next project with $300 in free Google Cloud credits—no strings attached. Test, build, and deploy without risk. Use your credits across the entire Google Cloud platform to find what works best for your needs. After your credits are used, continue with always-free tier services. Only pay when you're ready to scale. Sign up in minutes and start exploring.
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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 .
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VoiceOne gives you the ability to install and compleatly configure a pbx platform based on Asterisk 1.8 with an easy to use web GUI, which would be a framework to build a communication server adding various plugins.
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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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TurnServer is a implementation of Traversal Using Relay around NAT (TURN) protocol. This protocol allows a client to obtain IP addresses and ports from such a relay.
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Our Free Plans just got better! | Auth0You asked, we delivered! Auth0 is excited to expand our Free and Paid plans to include more options so you can focus on building, deploying, and scaling applications without having to worry about your security. Auth0 now, thank yourself later.
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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.
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ICTInvoice
A complete invoice managment solution for Elastix PBX
...It automatically create pdf invoices and emails to Elastix users on monthly basis along with their calls detail records ICTInvoice Main Features Multitenant Create invoices for Multiple campanies with branding Manual as well as Automatic invoice generation on fix date Auto Emailing invoices to users on monthly basis Inbound Billing Outbound Billing Extension / User wise reporting Summrize reporting Detail Reporting For more information , please visit http://www.ictinnovations.com -
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VozBox. Más que una centralita.
VozBox. Asterisk VoIP PBX.
VozBox es poder disponer de un Sistema de Comunicación Avanzado. Es poder llegar más lejos. Es poder convertir lo más sofisticado en lo más sencillo. Es poder contar con la más avanzada tecnología de comunicación, reduciendo sus gastos. 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 -... -
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mod_MumbleLink
Positional Audio Communication for Minecraft with Mumble
A Mod so that Minecraft now natively supports Mumble's positional audio feature. This means: Directional and positionally attenuated VOIP in relation to the game world. Please visit the Forum for information about the newest Version! Main Forum-Thread: http://www.minecraftforum.net/forums/mapping-and-modding/minecraft-mods/1272675 TheSkorm's Fork on GitHub: https://github.com/TheSkorm/mod_mumblelink Mumble: http://mumble.sourceforge.net Donations are greatly appreciated: https://sourceforge.net/donate/?user_id=2370023 -
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Open source callcenter suite focused on high performance and easy administration.
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vTiger Freeswitch PBX Integration NYFON
vTiger Freeswitch PBX Integration
vTiger Freeswitch Integration by NYFON. Allows to perform outbound (Click to Call) and incoming (wip) calls from vTiger 6.1+ interface. Modified PBXManager allows to choose between Asterisk and Freeswitch for PBX integration. Same (and extended in the future) functionality as Asterisk interface. -
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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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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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This application use sipp. The GUI base application try to create xml and csv files easily and start scenario which are selected. sipp and mono have to be installed on your PC. If you want to send RTP packets, you should copy pcap files to same folder where running sipp_gui.
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SIP Anonymization Tool (SiAnTo)
Small and effective program for SIP traces anonymization
...Our aim is to promote analysis of real SIP-server behavior. Therefore, we present SiAnTo, an extended anonymization technique that substitutes session-participant information with matching, but nondescript, labels. This allows for SIP traces to be publicly shared, while keeping interesting traffic-session properties intact. -
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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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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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An open source software project delivering converged voice and data applications. The StarPound platform enables you to rapidly develop voice and data solutions using the power of business process modeling. More info: http://www.starpound.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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asterCRM is a call center software for asterisk based VoIP system, also it has some CRM functions. It provide useful features such as pop-up, predictive dialer, click to call, extension status .... astercrm could work with all asterisk based system.
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ACR - Asterisk CDR Reporter
Information on Asterisk cdr like ASR, ACD, Billing duration 'billsec',
Information on Asterisk cdr like ASR, ACD, Billing duration 'billsec', time criteria not only date, I will add more features once I have time, like channels usage and time zone shift. you are welcome to suggest ! Please feel free to download, install and/or modify the code and let me know your feedback. You can use the following CLI commands to download and install the script: cd /var/www/html wget http://citylan.dl.sourceforge.net/project/acrasteriskcdrr/rpt.tar.gz tar xzvf rpt.tar.gz then point your internet browser to http://your-ip-address/cdr.htm
