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Showing 27 open source projects for "can"
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JDA
Java wrapper for the popular chat & VOIP service
...After setting the token and other options via setters, the JDA Object is then created by calling the build() method. When build() returns, JDA might not have finished starting up. However, you can use await ready() on the JDA object to ensure that the entire cache is loaded before proceeding. -
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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. -
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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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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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Build Agents and Models on One PlatformGemini Enterprise Agent Platform is Google Cloud's comprehensive platform for developers to build, scale, govern, and optimize agents and models. Choose from Google's most advanced models and third-party models like Anthropic's Claude Model Family.
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AVAYA CMS API Server
CMS API Server with web site
Free version of AVREMO.CMS.API exposing REST Services. With this software you can browse for all CMS reports. Design a report with AVAYA CMS Supervisor designer, application automatically expose data as XML or JSON. You can also browse reports with your preferred browser (desktop or mobile) and execute CMS operations like modify dictionaries, modify agent skills, modify VDN/Vector associations and more. Application package include : - A responsive WEB server interface (compatible with mobile browsers) to browse reports, perform CMS Operations and Agent Skill modification from any device having a browser... -
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NoiseGator (Noise Gate)
A simple noise gate app intended for use with VOIPs like Skype.
...Mac users can use SoundFlower. -
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Eternity smdr
perl based SMDR/CDR capture & display on Web based PHP with mysql DB
You can Capture Matrix Comsec' Eternity/NAVAN product Online SMDR/CDR using Perl Script. Store it in MYSQL Database and display it by using filter in PHP -
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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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mumb1e
A Murmur-Server Webinterface written in PHP for Mumble
Administration Tool for Mumble -/- Murmur Servers. Mumb1e Admin Plugin, also named "MAP" is a powerful Murmur web interface application written with PHP that can create useres, edit themself and a lot more for Mumble server. MAP is developed over m -
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ictbilling
A client management, billing and e-commerce solution
ICTBilling contains a set of Drupal 7 modules that can be used with Ubercart E-commerce solution and ICTBroadcast, a unified communication auto dialer software. This Client and Billing solution for ITSP has following modules and rules: client module, ictservice module, ratelist module and rules for ubercart. Client module handles clients user interfaces, subscribed service details, service links, list of available addons and current balance. -
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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 -
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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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Zanzibar is a complete, standards based IVR. It includes an MRCPv2 Server with ASR and TTS engines as well as an voiceXML interpreter so that you can deploy and run voiceXML applications. It integrates with VOIP PBX’s (like Asterisk) using SIP and RTP.
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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
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SIP Inspector is a tool written in JAVA to simulate different SIP messages and scenarios. You can create your own SIP signaling scenarios, customize SIP messages and monitor incoming and outgoing messages. The tool can play RTP streams from a pcap fi
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The web application offers to import call protocols of phone calls and lists of billed calls from different telephone providers. They can be matched to existing call protocol entries to generate cross-provider phone bills.
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The project "Sippie" is an additional tool for automatically converting Wireshark-Traces to XML - scenarios, which can be easily used as infile for the OpenSource SIPp test tool and traffic generator.
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miTester for SIP is an automated SIP testing tool designed and developed to take care of the complex pre-deployment testing of SIP applications easily. This SIP testing tool can be used to simulate SIP call-flows & automate functional, regression tests.
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Astmontray is a systray applet written in PyQt so users can monitor whether their PABX servers running Asterisk.org, the open source telephony platform, are up or not. It supports AMI authentication and runs fine on Linux, OSX and Maemo tablets.
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A Sip live audio feeding agent. The agent captures audio from sound card and sends live audio stream(uLaw) to caller(sip phone) using RTP. It is based on Peers 0.3(http://peers.sourceforge.net/). Can be used in IP telephony to broadcast live audio.
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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.
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Visual voicemail viewer for Asterisk/AsteriskNOW written in PHP. Users log in with their extension and v/m password and can download messages with the web browser.
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FATS - FATS is a Twisted and Fast Asterisk's Telephony Services. Project contains implementation of FastAGI, AMI protocols for the Twisted framework. Using it you can develop fast and pretty services for the Asterisk IP-PBX.
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AS3 skype API for the Skype. Flash has access to the system Webcam, Audio and flash 9+ has all the capabilities to build a Socket communications. So we can have a Flash based skype client on the web. This can be a widget for instant chat & voice call.