Search Results for "gnu/linux" - Page 2
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Showing 257 open source projects for "gnu/linux"
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AstCDR
AstCDR
Sistema de CDR para Asterisk Puro em Portugues do Brasil. -
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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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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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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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SkypeInYourLanguage
Skype language files for additonal languages (details below)
...I will back up the current files and then shut down the project at the end of the month. Please make your own backups from SourceForge because I till take that down too. If anyone wishes to take it on to handle just the Linux part, please give me a shout. Thank you for all your hard work over the years and really sorry to see this go but it's probably time to find an alternative to Skype that's localizable and OS. -
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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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...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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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. . -
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Callflow Sequence Diagram Generator
Callflow Sequence Diagram Generator
The callflow sequence diagram generator is a collection of awk and shell scripts that will take a packet capture file that can be read by wireshark and produce a time sequence diagram. This is useful to view & debug SIP callflows or other network traffic -
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RADIUS to OSP
Connverts RADIUS acccounting messages to ETSI OSP CDRs
The RADIUS to OSP project is a module for the FreeRADIUS server which converts RADIUS accounting records into ETSI OSP Usage Indication messages. -
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A Web Control Panel Application for the OpenSIPS, which is intended for both system and user provisioning. It features more than 18 tools, covering important functionalities (MI,statistics) and modules (acc,siptrace,drouting,dialplan) of OpenSIPS. IMPORTANT: this is no longer the main hosting for the project. This was moved on GITHUB - https://github.com/OpenSIPS/opensips-cp .
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FreePBX (formerly Asterisk Management Portal) is a project to bring together best-of-breed applications to produce a standardized implementation of Asterisk complete with web-based administrative interface.
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VoiceOne gives you the ability to install and compleatly configure a pbx platform based on Asterisk 1.8 with an easy to use web GUI, which would be a framework to build a communication server adding various plugins.
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Asterisk Monitor is a HTML interface that acts a operator pannel for asterisk to display user/peer status and calls. This uses a reverse AJAX, PHP and Python to originate, transfer and hangup calls, manage queues and meetme rooms.
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PolycomVVXControl
A command line utility for remote controlling Polycom VVX IP phones
Application to remote control Polycom VVX IP phones via their web interface (using HTTPS). This application is initially intended to perform certain actions on phones running in Microsoft Lync mode. These actions include: * get device information * get status * sign in using PIN authentication * sign out * reboot * factory reset It also supports performing actions in batch reading data from a CSV file. -
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Avapbx
A customized Asterisk with support for Farsi language
این پروژه تغییر نام یافته است و با نام آواسینتا ادامه می یابد. برای اطلاعات بیشتر به صفحه اصلی آواسینتا در شبکه پلاس مراجعه نمایید. https://shabakehplus.com/projects/avasinta/ -
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jpbxlite
Java VoIP/SIP PBX system (replaced by jfPBX)
jPBXLite is a VoIP/SIP PBX. Supports SIP extensions, voicemail, trunks, conferences, queues (ACD) and an IVR system. Support video conferencing with jPhoneLite/1.4.0. NOTE:THIS PROJECT WAS RENAMED AND IS NOW jfPBX. Please go to jfpbx.sourceforge.net -
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Asterisk CDR import
Simple web interface for mapping CDR from VoIP provider to Asterisk
This is a simple tool to map information for the cost of phone calls given by your VoIP provider, to the call log inside your Asterisk. As a result, you will know how much each of your employees has spent for phone calls. Additionally, this tool shows information about the assigned DID and CLIP numbers in Asterisk. This is helpful to identify obsolete, duplicate and multiple assignments. -
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TurnServer is a implementation of Traversal Using Relay around NAT (TURN) protocol. This protocol allows a client to obtain IP addresses and ports from such a relay.
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MumPI - Mumble PHP Interface
An administration web interface for Mumble VoIP servers in PHP
The Mumble PHP Interface (MumPI for short) allows you to manage your Mumble servers via a webinterface. Users can register, upload textures, and retrieve account data. Admins can manage virtual servers, registrations, admin accounts, online users…
