Search Results for "call"
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Showing 26 open source projects for "call"
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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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Activa brings the Asterisk IP PBX to the call center. Built on top of Asterisk, Activa components enable successful call center implementations adding value in areas such as computer telephony, screenpop&click2dial, agent control, automatic dialing...
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GNU Gatekeeper (GnuGk)
H.323 Gatekeeper for VoIP and videconferencing
...Please find current versions at https://www.gnugk.org/ The GNU Gatekeeper (GnuGk) is a full featured H.323 gatekeeper under GPL license. It supports VoIP and videoconferencing and includes Radius and database support and many call routing functions. The project has moved! Please find current versions at https://www.gnugk.org/ -
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cyber-wrt PBX
Cyber-wrt pbx is embedded system in DD-Wrt router
...It's first an original IPBX GUI with all features needed for company PBX. it include if you need a Hotel GUI offering what a Hotel PBX must do billing, check-in check-out, wake-up, housekeeping, direct call in room and many more. It include also a captive portal system for internet access controle using one lan port on your router. Try it, you will not belie -
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OSDial
OpenSource Call Center and Dialer System
An Open Source (predictive) Dialer. OSDial is a full featured GPL dialer which does Manual, Power and Predictive dialing out of the box. Its easily installed through one of our .iso images. Install manual available at http://store.osdial.com/ Full details available at web site (http://osdial.com). -
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Sipp is a performance testing tool for the SIP protocol. Its main features are basic SIPStone scenarios, TCP/UDP transport, customizable (XML-based) scenarios, dynamic adjustment of call-rate and a comprehensive set of real-time statistics.
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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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tcliaxc is a Tcl extension which provides you access to iax voip protocol via iaxclient library. tcliaxc lets you use a Tcl package to do basic iax operations such as registering to a server, make a call, handle iax event and so on.
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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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An Asterisk configuration/dial plan manager written Object Pascal (FreePascal/Freevision). Supports multiple trunks and call groups. This is a multi-Tenant system requiring very little configuration to have a fully working PBX system.
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Kiax is a softphone (soft phone, VoIP client) with a simple and comfortable user interface for making VoIP calls to Asterisk PBX. It depends on the iaxclient library to use Asterisk's IAX2 protocol for easy call configuration and audio setup.
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VoIP Toolkit / Call Control with Integrated Media. High-level Java API for creating SIP enabled VoIP applications. Suitable for either the desktop (softphone, phone applet, incoming call gatekeeper) or server-side (auto attendant, ACD, voicemail).
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Scx-tools is a set of utilities for Skype VoIP client. The project goal is to provide several useful tools for such task as Skype calls recording and management. Currently only the call recorder (version 0.1.1) is available.
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*starShop is a professional billing and monitoring system based on Asterisk PBX for call shops (taxiphone), Internet cafes, libraries, schools and any place that provide phone calls for public use. For the latest news visit tinyurl.com/2cfwnn
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SIPStir is a bulk call generator for SIP devices/servers. The server piece is capable of generating an infinite variety of call scenarios based on a simple scripting language. A wxPython client can connect to the server to monitor and modify progress.
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Enable Linux firewall to support connection tracking and NAT of H.323 protocol. It supports RAS, Fast Start, H.245 Tunnelling, Call Forwarding, Signal Proxy/Softswitch, RTP/RTCP and T.120 based audio, video, fax, chat, whiteboard, file transfer, etc.
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Skype-based wardialer written in C#, can record call results, play a wav file upon the phone being answered. Does full seqential dialing. Saves results to log file.
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GibPhone is a highly extensible VoIP/IM client for the .NET framework that uses a powerful plugin engine to allow for UI extensions and any call control stack / media payload / transport protocol combination.
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JAGIServer is an Open Source, Multi-Threaded 100% Pure Java AGI Server for use with the Asterisk PBX (TM), providing full AGI call handling functionality. THIS PROJECT HAS BEEN REPLACED BY ORDERLYCALLS
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GkAuth is a call management system written in PHP and Perl for the GNU Gatekeeper project. It is designed as an external application
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Open source Call Center software, based on H.323/SIP protocols. Contains Nauphone(Software multi-channel IP Phone, with conferences and ability to integrate with other systems(CRM)) and NauLib(VoIP Library based on OpenH323 and Vovida SIP)
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H323 Auto Caller is a simple H323-based WAV file player: you use it to call a H.323 telephone and it will play a WAV file to the remote telephone over the H.323 connection. Possible uses include Voice alerts with Nagios/OpenNMS and hotel wakeup systems
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Open H.323 Call Generator is an OpenH323 (www.openh323.org) spinoff project for testing and performance analysis in H.323 environments.
