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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.
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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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
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.
Tipic Inc. is proud to announce the born of the first Open Source, P2P, VoIP and Video conferencing solution for XMPP/Jabber. The target of this project is to develop a new portable set of codes using high quality libraries to add VoIP and Video to XMPP
JabbahVoice is an open-source framework which allows developers to easily create and extend voice portals using Asterisk (www.asterisk.org), the famous open-source IP PBX.