Search Results for "gnu linux" - Page 2
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Showing 762 open source projects for "gnu linux"
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openLTE
An open source 3GPP LTE implementation.
OpenLTE is an open source implementation of the 3GPP LTE specifications. Currently, octave code is available for test and simulation of downlink transmit and receive functionality and uplink PRACH transmit and receive functionality. In addition, GNU Radio applications are available for downlink transmit and receive to and from a file, downlink receive using rtl-sdr, HackRF, or USRP B2X0, LTE I/Q file recording using rtl-sdr, HackRF, or USRP B2X0, and a simple eNodeB using USRP B2X0. The... -
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open5g
open5g is an open source implementation of the 3GPP 5G specifications.
Currently an initial implementation of RRC and part of the PHY are available as a library and three applications are available: - GSCN File Recorder - Records I/Q files based on the synchronization raster. - NR-ARFCN File Recorder - Records I/Q files based on the NR-ARFCN raster. - GSCN Scanner - Attempts to detect channels on the synchronization raster. -
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jphonelite
Java VoIP Softphone (SIP) (replaced by jfPhone)
jphonelite is a Java SIP VoIP SoftPhone for Desktops (Windows, Linux, Mac) and Android. Features 6 lines with transfer, hold, conference (up to all 6 lines), g711 u/a, g722, g729a, and video (video support in Linux or Windows only and includes H263/H264/VP8). Applet includes full JavaScript support. STUN/TURN/ICE supported. Encrypt media with SRTP. DTLS Key Exchange. Video conferencing with jPBXLite/0.12+ NOTE : THIS PROJECT HAS BEEN SUPERSEDED BY jfPhone. Please go to... -
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Modem Manager GUI
Frontend for ModemManager 0.6/0.7, Wader and oFono
Modem Manager GUI is a simple GTK based graphical interface compatible with Modem manager, Wader and oFono system services able to control EDGE/3G/4G broadband modem specific functions. You can check balance of your SIM card, send or receive SMS messages, control mobile traffic consuption and more using Modem Manager GUI. -
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Free Telephony Project
Free software and hardware for telephony
Legacy SVN repositories for projects from David Rowe http://rowetel.com Mailing lists for Codec 2 and SM1000. -
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jfBroadcast
VoIP/SIP Autodialer
VoIP/SIP AutoDialer. Broadcasts a message with the option to transfer person to another number. Now includes new survey options. -
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CAS OS
Linux operating system source distributions for the CAS product suite.
Provide and share included open source for the CAS product family SUSE Linux and CentOS OS distributions with the community. (Older versions are being moved here from other locations (DigitalCM)) -
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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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OpenSIPS (former OpenSER) is an GPL implementation of a multi-functionality SIP Server that targets to deliver a high-level technical solution (performance, security and quality) to be used in professional SIP server platforms. IMPORTANT: this is no longer the main hosting for the project. This was moved on GITHUB - https://github.com/OpenSIPS/opensips
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AI-powered service management for IT and enterprise teamsGive your IT, operations, and business teams the ability to deliver exceptional services—without the complexity. Maximize operational efficiency with refreshingly simple, AI-powered Freshservice.
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A project for a highly scalable protocol and framework and a collection of software tools for m2m (machine to machine communication) and IoT Internet of Things. The programs here works on Windows and Linux and are based around VSCP, The Very Simple Control Protocol. Repository is here: https://github.com/grodansparadis/vscp_software and here https://github.com/grodansparadis/vscp_firmware
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Telemarketing (Junk) Call Blocker
Blocks unwanted calls using caller ID and a modem
Program to block junk calls. Connects to a modem and listens for a caller ID. It then reads strings from a blacklist file and scans them against the caller ID. On a match, it sends commands to the modem that terminate the call. RECENT ACTIVITY: (11/2014):Added Raspberry Pi support for two USB modem brands. (01/2015): Added Raspberry Pi support for USR5686G modem. (06/2015): Added full-function program for ATian voice/FAX modem. (01/2016): Added jcEdit.py file edit program. (05/2016):... -
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Accent Tool Suite
Accent interfaces and controls systems through goals and policies
Accent (Advanced Component Control Enhancing Network Technologies) is a comprehensive tool suite that interfaces a variety of communications systems and allows these systems to be controlled through goals (high-level user aims) and policies (lower-level system rules). Accent has been applied to the domains of: o Call Control: for telecommunications, particularly call control in Internet telephony o Home Care: for home automation and telecare, particularly for domestic appliances... -
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oreka
Enterprise telephony recording and retrieval system
Enterprise telephony recording and retrieval system with web based user interface. The project currently supports recording voice from VoIP SIP, Cisco Skinny (aka SCCP), raw RTP and audio sound device and runs on multiple operating systems and database systems. It can record audio from most PBX and telephony systems such as BroadWorks, Metaswitch, Asterisk, FreeSwitch, OpenSIPS, Avaya, Nortel, Mitel, Siemens, Cisco Call Manager, Cosmocom, NEC, etc... It is amongst others being used in Call... -
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SNEP is an graphical user interface to manage Asterisk PBX. It provides high level control over routing and users administration making the PBX administration easier to non-technical users. Access our Chat community at Slack: http://snep.slack.com
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SIPSL (SIP Service Layer) is a Programmable Session Border Controller (SBC) and SIP B2BUA, as defined in RFC3261. Written in c++ and multithreaded. The Application logic can be implemented by extending the application class and implementing the call backs.
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GNU Gatekeeper (GnuGk)
H.323 Gatekeeper for VoIP and videconferencing
The project has moved! Please find current versions at https://www.gnugk.org/ The GNU Gatekeeper (GnuGk) is a full featured H.323 gatekeeper under GPL license. It supports VoIP and videoconferencing and includes Radius and database support and many call routing functions. The project has moved! Please find current versions at https://www.gnugk.org/ -
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asterisk-kannel
Asterisk and kannel integration
This project aims to integrate Asterisk IP-PBX with Kannel SMS gateway. -
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Intel Integrated Performace Primitives audio/video codecs plug-in for the OPAL/OpenH323 library including G.728, G.729, G.723.1, G.722.2 GSM-FR, GSM-AMR, H.261, H.263, H.264 and MPEG4 part 2.
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Taki
SIP softphone
Cross-platform SIP softphone -
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Vaak (Elastix-Persian)
http://www.vaakpbx.com
سيستم تلفني واك مبتني بر استريسك و محيط گرافيكي الستيكس، تنها سيستم تلفني ويپ فارسي دنيا . -
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GetSmart is a download manager that supports Multi-Connection downloads. With Multi-Mirror search it can download different parts from these mirror servers simultaneously. It can 'act as proxy' to serve multiple clients. It can also works as a Daemon.
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Vem ai o Novo Disc-OS Aguardem.. Uma Empresa Orkestral Estamos Voltando! Modificações: Asterisk 21 Nova Interface UI Centos 8 Nodejs Disc-OS é uma distribuição de um PABX IP baseado em software livre. Desenvolvido para o mercado brasileiro com interfaces em português, de fácil instalação e configuração, contendo Linux customizado, software Asterisk 1.4 e configurador Disc.
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PLEASE NOTE that all current AstLinux development is on GitHub: https://github.com/astlinux-project/astlinux Thanks to SourceForge for providing free hosting to the AstLinux Project for more than 10 years! Old source archive remains on SourceForge. AstLinux is a custom Linux distro centered around Asterisk, the Open Source PBX. AstLinux has many features that make it ideal for embedded and commercial Asterisk based solutions.
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Y.V.S-Bot
A Discord BOT using Discord.js
The BOT is used in our Discord server. -
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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. .
