Search Results for "web sip"
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Showing 57 open source projects for "web sip"
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Our computer monitoring software allows employees, field contractors, and freelancers to manually clock in when they begin working on an assignment. The application will take screenshots randomly or at set intervals, which allows employers to observe the work process. The application only tracks activity when the employee is clocked in. No spying, only transparency.
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Routr
The future of programmable SIP servers
Lightweight sip proxy, location server, and registrar that provides a reliable and scalable SIP infrastructure for telephony carriers, communication service providers, and integrators. Routr provides all the tools required to deploy your VoIP network, including a command-line for remote server control. It can also be controlled via a RESTful API or a web-based GUI. Routr includes all tools for deploying your VoIP network. It offers remote server control via command-line, RESTful API, or a user... -
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VoIP monitor
VoIP SIP and SKINNY quality analyzer and packet / audio recording tool
VoIPmonitor is open source network packet sniffer with commercial frontend for SIP SKINNY MGCP RTP and RTCP VoIP protocols running on linux. VoIPmonitor is designed to analyze quality of VoIP call based on network parameters - delay variation and packet loss according to ITU-T G.107 E-model which predicts quality on MOS scale. Calls with all relevant statistics are saved to MySQL or ODBC database. Optionally each call can be saved to pcap file with either only SIP / SKINNY protocol or SIP/RTP... -
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*Until you install the operating system, the default user = live / no password. DragonOS FocalX (22.04) and DragonOS Focal (20.04) are out-of-the-box Lubuntu based x86_64 operating systems for anyone interested in software defined radios. All source installed software is located in the /usr/src directory while the remaining software was installed by package managers. What is DragonOS and why do you want it? The shortest distance between two points is a straight line. DragonOS...
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A C++ framework utilizing Design Patterns for creating Linux and Windows communications applications that contain Dialogic® products. Includes media and network classes (analog, digital, SIP, H323), multithreaded event handling, distributed app support.
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Comet is a flexible backup platform, giving you total control over your backup environment and storage destinations.
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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
VitalPBX is a free telephone and communications PBX system for companies. 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... -
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Cisco TMS PhoneBook - PHP Version
Emulates Cisco TMS PhoneBook for CMM or Endpoints with PHP
... new participants to a existing conference. The search is regular expression and the filter compare the user query with the endpoint name or H323/SIP dial string from the database. All you need is a Web server (tested with apache) with PHP and "RewriteEngine on" to make it work Optionally you can use a SQL server to have a real database, but I used a simple array as db. -
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TeeBX
Asterisk based VoIP pbx appliance
TeeBX wants to be a VoIP appliance with a user interface to easy manage a communication platform based on Asterisk (forked from askozia pbx). Website, support, documentation (coming soon): http://www.teebx.com/ -
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Xplico
Xplico is a Network Forensic Analysis Tool (NFAT)
Xplico is a Network Forensic Analysis Tool (NFAT). The goal of Xplico is extract from an internet traffic capture the applications data contained. For example, from a pcap file Xplico extracts each email (POP, IMAP, and SMTP protocols), all HTTP contents, each VoIP call (SIP, MGCP, MEGACO, RTP), IRC, WhatsApp... Xplico is able to classify more than 140 (application) protocols. Xplico cam be used as sniffer-decoder if used in "live mode" or in conjunction with netsniff-ng. Xplico is used... -
Save time, money and your sanity. Deluxe Payment Exchange+ (DPX+) is our integrated payments solution that streamlines and automates your accounts payable (AP) disbursements. DPX+ ensures secure payments and offers suppliers alternate ways to receive funds, including mailed checks, ACH, virtual credit cards, debit cards, or eCheck payments. By simply integrating with your existing accounting software like QuickBooks®, you’ll implement efficient payment solutions for AP with ease—without costly development fees or untimely delays.
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Vodia PBX
A Better PBX For Your Business
Vodia PBX has included a complete suite of business telephony features for on-premise and cloud-based telephony deployments. Sign up at https://vodia.com/en/register and receive your free Vodia license. -
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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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Andspoilt
Run interactive android exploits in linux.
Andspoilt is a command line user interface designed to easily exploit android devices. Run interactive android exploits in linux by giving the users easy interface to exploit android devices Uses an intergration with Metaspoilt Framework by giving the user an easy interface to create payloads and launch android exploits. Current Additional feature is a simple web server for file distribution. -
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TOFFEE-DataCenter
Linux Server and Data-Center WAN Optimization
TOFFEE-DataCenter is the new TOFFEE variant meant for Data-Center, Server, HPC, Load Balancing, Cluster/Cloud Computing and SDN deployments. With TOFFEE-DataCenter (and or TOFFEE), OEMs and other equipment vendors can make use of this platform/stack, integrate, port and build WAN Optimization devices/appliances on to their existing product range. Or build a dedicated WAN Optimization hardware product series via the same. -
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Zen Load Balancer becomes Zevenet
becomes ZEVENET
High performance and easy to use open source load balancer with enterprise solutions. -
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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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Asterweb is an Asterisk Realtime Configuration utility written in PHP. It configures the realtime settings for voicemail, extensions and sip buddies. Users can login with their voicemail user and pin and check their voicemail.
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Mobicents is the leading Open Source VoIP Platform. It is the First and Only Open Source Certified implementation of JSLEE 1.1 (JSR 240), and SIP Servlets 1.1 (JSR 289). Mobicents also includes a powerful and extensible Media Server.
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Steve Murray Phone
simple SIP video phone source/lib/dll/API with h264, ulaw and g729
... someone needs a SIP video phone. It has been used to make calls as an ActiveX plugin, but it is pretty easy to put inside a Mozilla/Chrome plugin to make browser-based calls from a web page as well. You can drag-drop it as a control into any Windows program in Visual Studio. Do what you want with it, no need to bother me about it. -
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Git repo: https://github.com/asipto/siremis Web management interface for Kamailio (OpenSER) - handle subscriber profiles, access control lists, accounting, least cost routing and load balancing, monitoring charts, xmlrpc communication with SIP server
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CafeSip - Look what Java and SIP can do
A suite of open-source tools and frameworks for creating SIP apps
Session Initialtion Protocol (SIP) is widely used for telephone services over the Internet. CafeSip provides a suite of open-source tools and applications for creating customized SIP services and applications using the Java. -
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SIP UMS is an Unified Messaging Server for the SEMS/IVR. It is an extendable voicemai and IVR platform written in Perl. It includes a web based adminstration tool, voice prompts (in english) and sample scripts that will work with SEMS/IVR.
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In this project a real time VoIP signaling analyzer and alarming system was developed which deals with SIP signaling. . The popular C library WinPcap was used to build up the program. The packets were stored in MySQL database through MySQL shell.
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The Milkfish is an Embedded Communications Software Project. Main objective is to provide a basic NAT traversal solution for cheap hardware enabling a practical setup of multiple SIP phones in a LAN sharing a WAN connection with one public IP address.
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DragonEyes
An application for multiparty video/audio conference
The DragonEyes is a linux software which enables multimedia, multiparty collaboration in applications such as group conferencing, distance learning, training and video telephony. All written in C++ and with a GUI frontend. The interface is based on Qt. The media backend is based on the GStreamer framework. Communication protocols include SIP, RTP.