Search Results for "gnu assembler windows" - Page 3
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Showing 90 open source projects for "gnu assembler windows"
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Jabbin is an Open Source social application that combines VoIP, Instant Messaging and Social Networking, allowing you to focus on what you really care about: your friends.
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jSipgateConnect library allows Java-based access to all functions specified in the 'Sip Advanced Messaging, Usability, Routing and Accounting Interface' specification.
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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 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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Plug and play provisioning server for snom VoIP phones written in Python. The server supports the plug and play provisioning feature embedded in snom phones of the snom 3xx, snom8xx and snom MeetingPoint series.
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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 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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SerMyAdmin
Web interface for OpenSIPS
I'm sad to say that, unfortunately, the serMyAdmin project is being discontinued starting in September 1st, 2010. After three years of development, the project no longer makes sense. The main reason to create the serMyAdmin was the lack of a GUI for OpenSER/OpenSIPS. The OpenSIPS control panel fulfill this need and it is not wise to maintain two GUI projects. Our team will support the opensips-cp initiative to avoid splitting the community between two open source projects for OpenSIPS GUI. I... -
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A simple lightweight instant messaging and VoIP client wirtten in Java. It implements the SIP protocol as described in RFC 3621 and other related RFCs and implementation requierements.
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This is a SIP signaling layer to create a fully operative multipoint (video) conference server using SIP clients and RTP media streams in combination with strManager as a media management layer.
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Experimental open source project, which aims to test possibility of VoIP calls "from browser to browser". Main part is Java application implementing UDP hole punching, which is sent to both clients using JWS. See it online in action at project homepage
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VoIP's project with Speex codec, speech detecting and crypto coding with traffic counting. All code is a pure Java, so completely cross-platforms. Traffic is about 9 MB per hour in both ends, sources: http://www.open-source-soft.narod.ru/arrow.7z
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This project provides a meter of QoS on VoIP using the protocol SIP Media Loopback
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QoS Number is a project to control de quality of service ( QoS ) in a IP-Centrex Plattform .
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An XCAP server is used by XCAP clients to store data like buddy lists and presence policy in combination with a SIP Presence server that supports PUBLISH, SUBSCRIBE and NOTIFY methods to provide a complete SIP SIMPLE server solution.
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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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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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Clock in/out from any phone, or from the web portal. Comes with full web portal and reports. Get the latest build from SVN Checkout (Public/SVN Repository). For more information on Asterisk PHP Timeclock, visit http://www.asterisktimeclock.org
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DiWA (Dialplan Web Administrator) is a php/ajax interactive web application to manage Asterisk's diaplan. It supports both static and realtime configuration, and doesn't require database.
