Search Results for "asterisk-oh323"
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Showing 15 open source projects for "asterisk-oh323"
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Top-Rated Free CRM SoftwareHubSpot is an AI-powered customer platform with all the software, integrations, and resources you need to connect your marketing, sales, and customer service. HubSpot's connected platform enables you to grow your business faster by focusing on what matters most: your customers.
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AppKonference , a high-performance Asterisk conferencing module, is a fork of AppConference focused on voice.
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Replacement for the SCCP channel driver in Asterisk. Extended features include Shared Lines, Presence / BLF, customizable Feature Buttons, and Custom Device State. Visit our discussion mailing list for help and join us as a developer if you like. The project moved to https://github.com/chan-sccp/chan-sccp
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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. . -
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Web-MeetMe is a suite of PHP pages to allow for scheduling and managing conferences on an Asterisk PBX.
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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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8ix Zenith CE spells Asterisk derived IP Telephony solution.When you want robust communication, without the huge cost and recurring bill that comes with the way you do business, 8ix Zenith CE addresses your every communication need.
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A lightweight cross platform IP telephony client using the IAX protocol, designed for use with the asterisk open source PBX.
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HATEFNA هاتفنا .....http://www.hatefna.org ..... (PC&Mobile)IP SoftPhone over SIP and ASTERISK using JAVA for PC and Symbian C++ for Symbian Mobiles Phones ..... برمجيات للمهاتفة ببروتوكول الإنترنت
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User Testing Platform | TesteumTired of bugs and poor UX going unnoticed despite thorough internal testing? Testeum is the SaaS crowdtesting platform that connects mobile and web app creators with carefully selected testers based on your criteria.
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It´s a complete All-in-one suite of Asterisk PBX, Asterisk Managment Portal (AMP), Flash Operator Panel, Asterisk Recording Interface, Sugar CRM and Integrated WebMeetMe GUI.
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Jabast connects XMPP Clients to a asterisk PBX to allow extension listening, calls and presence controle for Agents.
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'The Genie' is a Java based SIP User Agent that can talk to any other SIP User Agent via SIP Proxy or Asterisk. Also Planned: 1-click sign-in to Yahoo or any other site by storing hashes of user/passwd pairs in a 'Wallet' and they can be accessed onl
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Utilities enhance AMP, A@H, Asterisk and Cisco XML. Speed Dialing revised using web interface. Paging using custom meetme routines. Intercom by dialing 0 + Ext #. Browsing AMP's extensions on your phones LCD. Backup patch for Asterisk@Home maint screen.
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A set of JNI classes providing direct access to Asterisk PBX functionality from Java. This is not a socket-level interface into the Asterisk manager app but a true Java<->Asterisk integration at the Thread level.
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The G.O.N.E. is a softphone (or soft phone) running over the web, fully multi-plataform, it implements the SIP protocol, and is built to work on any SIP server, like Asterisk, and others. GONE will work on a complex sistem, but this will be showed a bit
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