Search Results for "libamd.so.1"
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Showing 45 open source projects for "libamd.so.1"
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Train ML Models With SQL You Already KnowBuild and deploy ML models using familiar SQL. Automate data prep with built-in Gemini. Query 1 TB and store 10 GB free monthly.
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Synapse
Matrix reference homeserver
Matrix is an ambitious new ecosystem for open federated Instant Messaging and VoIP. Everything in Matrix happens in a room. Rooms are distributed and do not exist on any single server. Rooms can be located using convenience aliases like #matrix:matrix.org or #test:localhost:8448. Synapse is currently in rapid development, but as of version 0.5 we believe it is sufficiently stable to be run as an internet-facing service for real usage! Create and manage fully distributed chat rooms with no... -
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BlackBelt WASTE- ipv4 Tor i2p AI Voice
Modern, AI-Smart, WASTE p2p for ipv4, Tor and i2p + Voice Conference.
Open Source - GPLv3 inc images. A WASTE client. Download and create your own WASTE networks. Move 1000's of GB's at 100MB+ per sec. (800 Mbits per sec) FULL pause and resume capable. Voice Conference, Chat, Transfer files and Participate in Forums in a secure environment. For Windows XP 32/64, Vista 32/64, Win7 32/64, Win8 32/64, Win 10, Win 11, Linux (WINE). *** User Based Access Control - for voice, chats, file transfers and uploads. (useful in NULLNETS) *** Distributed... -
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Elastix
Unified Communications Server
Elastix is a software-based PBX powered by 3CX and based on Debian. An open-standards solution, Elastix is an easy to install and manage UC system compatible with popular IP phones, gateways and SIP trunks. Elastix is complete with unified communications features such as integrated WebRTC video conferencing, chat, presence and softphones and smartphone clients for Windows, Mac, iOS and Android. -
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VitalPBX
Unified Communications System - PBX System
...It is a complete platform that can be installed on the physical hardware on the site or as a hosted application. VitalPBX acts as the upper layer interface for the Linux base and then Asterisk (one of the most popular communication toolkits in the world). For this reason, VitalPBX is the graphic user interface between you and the complex world of modern communications. VitalPBX will help you implement a secure telephone system for your company, save, take advantage of recent innovations and provide opportunities to integrate your business processes if you wish to do so. -
$300 Free Credits to Build on Google CloudStart your next project with $300 in free Google Cloud credit. Spin up VMs, run containers, query petabytes in BigQuery, or build agents with Gemini Enterprise Agent Platform. Once your credits are used, keep building with 20+ always-free tier products including Compute Engine, Cloud Storage, GKE, and Cloud Run functions. No commitment required—just sign up and start building.
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Stuntman - STUN server and client
High performance, production quality STUN server and client library
New version 1.2. This is the code to STUNTMAN - an open source STUN server and client code by john selbie. Compliant with the latest RFCs including 5389, 5769, and 5780. Also includes backwards compatibility for RFC 3489. ICE and WebRTC ready. Version 1.2 compiles on Linux, MacOS, BSD, and Solaris. Supports the STUN protocol on both UDP and TCP for both IPv4 and IPv6. Windows binaries are also provided. Additional features are in development. This is a mirror of the code on... -
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pcapsipdump is libpcap-based SIP sniffer with per-call sorting capabilities. It writes SIP/RTP sessions to disk in a same format, as "tcpdump -w", but one file per SIP session (even if there is thousands of concurrent SIP sessions). Getting started: http://pcapsipdump.sf.net/
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Avasinta
آستریسک با پشتیبانی از زبان فارسی
آواسینتا ویرایشی اختصاصی شده از نرم افزار ویپ آستریسک می باشد که از زبان فارسی نیز پشتیبانی می نماید. این برنامه رایگان بوده و روی سیستم عامل اوبونتو ارائه می شود. برای دریافت اطلاعات بیشتر د ر مورد سیستم تلفن آواسینتا می توانید به صفحه اصلی آواسینتا در شبکه پلاس مراجعه نمایید https://samiyacode.ir/avasinta/ -
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integral
Fast CGI-based AsteriskPBX cluster administration tool.
Allows OnLine monitoring of all kind Asterisk functioality: - Real-Time channel monitoring; - Queue Agents control; - Conference Administration; - Config edition; - Asterisk Core monotoring; -
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VozBox. Más que una centralita.
VozBox. Asterisk VoIP PBX.
...Última versión disponible 2.11.19 en la web de VozBox: http://www.vozbox.es/descarga/ Internet connection required during installation. Update version: 2.11.19 1 - Customizing bash_login and bashrc 2 - Upgrade to version libpri-1.4.15 3 - Included new module-manager FOP2-1.0.3 4 - Inclusion of new packages and dependencies. 5 - Correction of errors in the installation. 6 - Upgrade to version webmin-1.710 7 - Correction of errors in the pbx-vpn script. 8- Inclusion of new script pbx-status. -
Build Agents and Models on One PlatformGemini Enterprise Agent Platform is Google Cloud's comprehensive platform for developers to build, scale, govern, and optimize agents and models. Choose from Google's most advanced models and third-party models like Anthropic's Claude Model Family.
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cyber-wrt PBX
Cyber-wrt pbx is embedded system in DD-Wrt router
...it include if you need a Hotel GUI offering what a Hotel PBX must do billing, check-in check-out, wake-up, housekeeping, direct call in room and many more. It include also a captive portal system for internet access controle using one lan port on your router. Try it, you will not belie -
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OSDial
OpenSource Call Center and Dialer System
An Open Source (predictive) Dialer. OSDial is a full featured GPL dialer which does Manual, Power and Predictive dialing out of the box. Its easily installed through one of our .iso images. Install manual available at http://store.osdial.com/ Full details available at web site (http://osdial.com). -
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PHP 2 Way Webcam Video Chat
1 on 1 Webcam Videochat Script with P2P Support
This is a web based instant 1 on 1 private online video conferencing solution. It's a solution for conducting easy to setup face to face meetings without leaving your office or home. It's the easiest and most cost-effective way to meet somebody and discuss one on one, to make a video call just by providing a private room access link. -
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SMPTE 2022-1 library for VLC & co
Optimized and cross platform SMPTE 2022 FEC library in C, Python, Java
Project moved to GitHub. https://github.com/davidfischer-ch/smpte2022lib -
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JADfactory
Application to make JAD files
A simple application to make jad files -
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Root Lounge - Mumble Skin
A Skin for Mumble
Spice up your Mumble experience using this recreation of 4 different skins merged and adjusted into a new one. It is tested and running stable using Mumble 1.2.x. To install: Download, extract and follow the readme.txt contained in the archive. All credits go to the devs of the individual pieces including Mumble itself. -
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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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Universal Queue and Unified Communications core implementation upon the Platform as a Service paradigm. An abstraction layer for the hardware communications platform among a plugin based model brings independence from technology and media to the project
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[es ] Aplicación web central para particulares. [en ] Core web application for individuals. [fr] Application Web centrale pour les personnes. [pt] Central de aplicação web para indivíduos. [ar] المركزي تطبيق ويب للأفراد
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This is an initiative to assemble/develop the cheapest, easiest to setup, easiest to manage, scalable, Open Source, standards-based, wireless local do-it-yourself telephone company toolkit in the world.
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SIP.NET is SIP stack .NET library written in C#. SIP message parser realized on Deterministic Finite-state Machine (DFA parse all SIP message for one pass). DFA generates automatically from Augmented Backus–Naur Form from RFC3261.
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Live-cd com Servidor SIP baseado em opensips. O Servidor de Telefone Livre transformará um simples PC numa central SIP multidominio. Estamos construindo a documentacao do projeto na wiki do sistema trac.
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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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astrtr
Asterisk module to route traffic (very obsolete)
astrtr is a FreePBX module for Asterisk to route calls from one trunk to another. Makes Asterisk PBX a VoIP Switch as well. Includes multiple routing tables, max connections per trunks, some reports, CDRs, etc.