Search Results for "apache web server" - Page 2
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Showing 70 open source projects for "apache web server"
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AGIS stands for Asterisk Groupware Integration Server. It provides a SOAP backend for access and configuration of the Asterisk PBX. Further it implements a SOAP based Asterisk application for E-Groupware with management functions for users and admins.
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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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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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CafeSip - Look what Java and SIP can do
A suite of open-source tools and frameworks for creating SIP apps
Session Initialtion Protocol (SIP) is widely used for telephone services over the Internet. CafeSip provides a suite of open-source tools and applications for creating customized SIP services and applications using the Java. -
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Webmin - Dahdi HW module
This Webmin module aids in configuring the Dahdi interface to Asterisk
...The legacy interfaces can be FXS, FXO, ISDN BRI, T1 over USB, T1, E1 ISDN PRI and TDMoE (Dynamic Dahdi) ports. This Webmin module aids in configuring the legacy hardware devices to Asterisk PBX. Webmin was chosen because Webmin is a mature web based GUI platform that concentrates on the hardware side of a server. Webmin allows controlled secure root access to the hardware. Other web based GUIs concentrate on the software side of configuring Asterisk and do not provide root access directly. For the latest information on installing the module in FreePBX go to the Wiki - How to Install page -
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The ATSlog software provides a handy web-oriented interface for collecting, viewing and analysing calls for various types of PBX (Private Branch eXchange) models.
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SIP UMS is an Unified Messaging Server for the SEMS/IVR. It is an extendable voicemai and IVR platform written in Perl. It includes a web based adminstration tool, voice prompts (in english) and sample scripts that will work with SEMS/IVR.
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Pronounced "Open Spippy" (like "skippy" but with a "p" instead of a "k"): A reference implementation of the Session Peering Provisioning Protocol binding for SOAP being developed in the IETF's DRINKS Working Group.
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GoAutoDial is an all-in-one open source Vicidial administrator utility that enables you to manage your Vicidial - CentOS based server. Package includes Vicidial web admin, GoAutoDial admin, NEW AgentUI. For more info, please visit www.goautodial.com.
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It is an appliance software that integrates the best tools available for Asterisk-based PBXs into a single, easy-to-use interface. include astercrm astercc voicemail,fax,et
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The project is a preconfigured VoIP PBX VM Image based on Asterisk. But provides more advance features, such as LDAP integration, VQ monitoring via RTCP XR reports, LDAP user login, integrated billing system and telephone directory PDF generator.
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The Java Teamspeak 3 Server Query Software Development Kit is a Java Library that provides Classes to access the Teamspeak Server Query for easy integreating TSSQ into Java Desktop- or Web-Applications.
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The minimum hardware to have a portable IP-Phone. A VoIP telephone that is simple to assemble and an emulator for portability. Core runs under Linux. Modular functionalities: a display, keys (dialpad, DND,...), a web server. Free, simple and useful.
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MurMan, the Murmur Manager, is a user friendly, robust, and attractive web management interface for Murmur servers based on PHP, using Murmur's (server component for the Open-Source voice communication Mumble) Ice RCP.
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Add-on module to help with the deployment of SafiServer applications in Asterisk/FreePBX based systems. The module will allow administrators to reference Saflets (graphical call flow/IVR applications) using the FreePBX web interface.
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MumurWeb will be a comprehensive webinterface for Murmur servers based on PHP. Murmur is the server component for the Open-Source voice communication Mumble.
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Mongo has bundled Asterisk 1.4.13 + add-ons 1-4.4 + freePBX 2.3.1 + add-ons + sugarOS 4.5.1 + XAMPP 1.6.4 + MYSQL (with it's own sock) + PHP (part of XAMPP) + oscommerce
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Xen-Agi is an open source java library that allow you to build java applications that interact with the Asterisk Server for VOIP and PBX functionality. This project supports the FastAGI protocol exclusively and is inspired by the Asterisk-Java project.
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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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AdminsParadise VoIP and Fax server offers a FREE Enterprise grade clustered phone and fax solution for your network for ultimate reliability. Features Web based faxing, scheduled conferencing, voice-mail, roaming users, and much more. Runs Asterisk 1
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The Asterisk-Java package consists of a set of Java classes that allow you to easily build Java applications that interact with an Asterisk PBX Server. It supports the FastAGI protocol and the Manager API.
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This is a simple software, comprised of a server and a client, that can be used together with any IP-phone using the SIP protocol (TELE2 and Bredbandsbolget in sweden). It also retrives information, sush as name, adress, etc about the number from eniro.s
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Bi-directional SMS converter for mobile phones and PC. nVMGc converts messages from certain mobile phones SMS to text file, database rows, or to other mobile phones' format. Participations are required to make nVMGc supports more mobile phone types.
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Easy, automatic reporting for VoiceXML based applications in particular Genesys GVP. Provides automatic menu option history, dialog history and storage of application variables. Do in minutes what used to take weeks.
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RAMS is a java based server useful for managing inter-domain VoIP routing, called number translation and Call Detail Record collection. RAMS supports the European Telecommunications Standard Institute's OSP Peering protocol (ETSI TS 101 321).
