Search Results for "java projects on linux" - Page 3
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Showing 66 open source projects for "java projects on linux"
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Secure File Transfer for Windows with Cerberus by RedwoodCerberus supports unlimited users and connections on a single IP, with built-in encryption, 2FA, and a browser-based web client — all deployable in under 15 minutes with a 25-day free trial.
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Error to trace to log to deploy. One click. No SSH.AppSignal links every error to the trace, the trace to the log, the log to the deploy that shipped it.
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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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Asterisk based callcenter system. Administration interface to manage queues, agents, accounts, skills - track calls, create statistics. Realtime queue monitor. Agent web interface with build in IAX softphone.
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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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This is a repository for QueueMetrics add-ons that are contribuited by the community and are open-source.
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Go from Code to Production URL in SecondsSkip the Kubernetes configs. Cloud Run handles HTTPS, scaling, and infrastructure automatically. Two million requests free per month.
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Olyo is an open source project which aims at implementing a prototype for Peer-to-Peer Session Initiation Protocol (P2PSIP) and providing real-time multimedia services over Internet based on this prototype. Olyo is promoted by MINE lab, BUPT.
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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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SipManager is a full-feature IP-Centrex PBX web app for hosted/multi-client PBX based on Asterisk and OpenSer. VoIP/ToIP platform (PBX, Fax, workflows, etc.). User-friendly web interface. Realtime architecture with SQL database. ASP. http://www.ovvoe.com
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SIP to Skype Gateway/Bridge/Converter/Adapter. Make and receive SIP to Skype Calls and Skype to SIP Calls. Call Skype users using speed dial or use SkypeOut. Make SIP calls from Skype using a SIP provider or SIP PBX. Use as a Skype Trunk with a PBX.
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Sipana is a distributed SIP monitoring tool to monitor the SIP signaling behavior using sequence diagrams and providing SIP end-to-end performance metrics through a centralized Web interface. For more information please visit http://sipana.org/
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$300 Free Credits for Your Google Cloud ProjectsLaunch your next project with $300 in free Google Cloud credits—no strings attached. Test, build, and deploy without risk. Use your credits across the entire Google Cloud platform to find what works best for your needs. After your credits are used, continue with always-free tier services. Only pay when you're ready to scale. Sign up in minutes and start exploring.
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Spark plugin for integratión with CentricCRM. This plugin for Spark XMPP client complements the integration between Asterisk PBX, Centric CRM and XMPP Server. Automatically open session on Centric CRM and redirect to contact the page
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A complete contact center solution for the Asterisk PBX, capable of handling inbound and outbound campaigns. Provides an agent toolbar for CTI. For managers an admin panel controls every aspect of the call center. Stunning historical and realtime stats.
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Present is an attendance dialer for educational organizations. It currently will integrate directly into Powerschool SIS, with the potential to connect to other SIS systems as well. It can be used to make automated attendance and tardy calls
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sippet
C++ SIP stack based on Chrome source code
This C++ library has been designed as a Chrome SIP stack. Sippet is an open-source SIP User-Agent library, compliant with the IETF RFC 3261 specification. It can be used as a building block for SIP client software for uses such as VoIP, IM, and many other real-time and P2P communication services. The main target was to enable Javascript applications to use UDP, TCP and TLS transports along WebSocket. Existing SIP solutions for the browser are forced to use the WebSockets API to...