Search Results for "ip-camera"
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Showing 71 open source projects for "ip-camera"
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Stop Cyber Threats with VM-Series Next-Gen Firewall on AzureGain integrated visibility across all traffic in a single pass. Deploy Palo Alto Networks VM-Series to determine application identity and content while automating security policy updates via rich APIs.
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Ship Agents FasterGemini Enterprise Agent Platform lets you rapidly build, scale, govern and optimize production-ready agents grounded in your organization's data. The platform enables developers to build custom or pre-built agents for virtually any use case. New customers get $300 in free credits.
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Siproxd is a proxy/masquerading daemon for the SIP protocol. It allows SIP clients (softphones & hardphones) to work behind an IP masquerading firewall or router.
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Open Phone Abstraction Library (OPAL) is a C++ multi-platform, multi-protocol library for Fax, Video & Voice over IP and other networks. Also included is the Portable Tool Library (PTLib) which is a C++ multi-platform abstraction library and collection o
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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... -
Our Free Plans just got better! | Auth0You asked, we delivered! Auth0 is excited to expand our Free and Paid plans to include more options so you can focus on building, deploying, and scaling applications without having to worry about your security. Auth0 now, thank yourself later.
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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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VMS1100, IPC1100
OSS code used in VMS1100 and IPC1100 set top boxes
Source code content for Open Source packages used by VMS1100, which is a Whole Home Video Gateway and the IPC1100 advanced IP Client video set-top -
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Kazoo Live Medium
Kazoo Live Medium. From Null to a Full Featured PBX on minutes
...Monster-UI Open Source Apps, like SmartPBX, CallFlows, PBX Connector, Voicemails, Faxes, Accounts and Number Manager Kazoo v4.3.x Kamailio v5.1 FreeSWITCH v1.8.x Bigcouch / Couchdb 2.2 RabbitMQ How to start up: - Burn the ISO image on USB Stick, or also on DVD - Boot it, choosing First boot option. - After starting up, login as root user, with password 'kazoo' - Execute 'install_kazoo' from bash shell. There are some config steps (hostname, IP Address. If you have issues starting kamailio, edit postgresql.conf after switching to postgresql user; set 'max_connections = 300' or higher value . -
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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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MagnusBilling
Free billing to Asterisk
MagnusBilling is a VoIP server system that brings together the best IP Billing Telephony software in the market, creating a comprehensive, flexible and superior tool. Enterprises that need the freedom to build their own VoIP server. You will be able to chose your termination, giving your business larger flexibility and autonomy. Many VoIP distributors use the best of each VoIP server, termination to be able to be more competitive and to have better price and quality. -
Atera - an All-in-one platform for IT managementYour IT essentials, integrated & elevated. Take your IT management from automated to autonomous, download Atera's agent to start your free trial!
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Activa brings the Asterisk IP PBX to the call center. Built on top of Asterisk, Activa components enable successful call center implementations adding value in areas such as computer telephony, screenpop&click2dial, agent control, automatic dialing...
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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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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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camerubuntu-phone-12.04
camerubuntu-phone-12.04 est un système ubuntu avec téléphonie IP.
camerubuntu-phone-12.04 est un système ubuntu intégrant la téléphonie IP. Avec ce système Linux, vous disposerez d'un serveur de téléphonie IP Asterisk orienté web qui vous permettra d'effectuer sur votre réseau des appels téléphoniques par ordinateurs, par téléphones IP et mếme en utilisant vos smartphones. Une fois installé, il suffit juste de connecter le PC au réseau, de créer les numéros en utilisant un navigateur, installer un client softphone comme 3CX ou autre dans les PC clients et vous pourrez passer des appels dans votre réseau de façon illimitée. -
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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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VIP28x3
IP Set-top box for broadcast HDTV, services, and more
The VIP2853 is a cost-effective IP set-top box that offers a flexible way of bringing high-quality IPTV services, such as high-definition broadcast TV, on-demand services, interactive applications, and DVR/time-shift, as well as over the top (OTT) services, to the home. -
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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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SIP Data Filter (SiDaFir)
Simple and efficient tool for SIP trace filtering
The Session Initiation Protocol (SIP) is a signaling communications protocol widely used nowadays for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks. Thanks to its simplicity, SIP messages are often used in creative ways for which these were not originally designed (e.g. using periodical OPTIONS packets as NAT keep-alive instead of using STUN or TURN) and thus SIP traces of the captured traffic often contain "useless" traffic from the actual signalization point of view. ... -
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PBXlab adds value to business thanks to its valuable contribution of technological solutions. PBXlab promotes the use of free software programs that can be tailored to the real needs of its users, speed of movement between users, at a very low cost or no cost. PBXlab offers development of add-ons and improvements to programs, leading to more full, safe and adapted to customer needs programs. We offer technological independence without having to develop products from...
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CASMR GIL
Source Code content for Open Source packages used by the CASMR GIL
Source Code content for Open Source packages used by the CASMR Gateway Image Loader (GIL), which is a CASMR Plug-in module that allows the deployment of code download campaigns to the VMS and IP Client devices. -
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CASMR CDL
Source Code content for Open Source packages used by the CASMR CDL
...Code Download signaling or provisioning instructions are delivered through the out-of band control channel for DCII terminals. Code objects are carouseled either on the out-of-band control channel, on a specified in-band stream, or defined IP Multicast. The provisioning commands provide enough information for the terminal to acquire the code download data stream, load, and then activate the code object. -
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VozBox. Más que una centralita.
VozBox. Asterisk VoIP PBX.
VozBox es poder disponer de un Sistema de Comunicación Avanzado. Es poder llegar más lejos. Es poder convertir lo más sofisticado en lo más sencillo. Es poder contar con la más avanzada tecnología de comunicación, reduciendo sus gastos. Es una centralita VoIP Asterisk con altas prestaciones para las Empresas. 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 -... -
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oSKY
Framework implementing SS7 over IP & SIGTRAN interworking and trunking
oSKY (OpenSKY) is a full communication framework to implement PSoTN (Public Social Telephony Network). In contrast with PSTN (Public Switched Telephony Network), oSKY intends to implement an open paradigm in the internet domain, gaining more than 100 years of experience from traditional telephone network. oSKY goals: -enhanced security (SSL, SCTP) -embedded NAT tunneling -Global title discovering on Internet style -Peer2Peer mode and Gateway mode -more... -
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SIP Anonymization Tool (SiAnTo)
Small and effective program for SIP traces anonymization
The Session Initiation Protocol (SIP) is a signaling communications protocol widely used nowadays for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks. A good way to design optimization techniques for SIP deployment would be to analyze SIP traffic from existing networks. However, publicly available analyses of SIP traffic are rare and thus not a lot of knowledge exists about typical behavior of a SIP server (as opposed to, for example, HTTP servers). Our aim is to promote analysis of real SIP-server behavior. ... -
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sigtran-smsc
PoC: Short Message Service Center (SMSC) over SIGTRAN
...It uses following protocol stack: MAP (moForwardSM and mtForwardSM) TCAP (Begin with Invoke + End with ReturnResultLast) SUA (SCCP User Adaptation): ASP UP/ASP ACTIVE + ACK's and CLDT messages only SCTP (Linux Kernel - SCTP implementation) IP Current implementation is tested in SAMK NGN [http://www.samk.fi/ngn] laboratory's 3G mobile network (real Nokia MSC/VLR network hardware with UTRAN RLC and node-B). Thesis URL: http://urn.fi/URN:NBN:fi:amk-2013052210011 -
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ACR - Asterisk CDR Reporter
Information on Asterisk cdr like ASR, ACD, Billing duration 'billsec',
...You can use the following CLI commands to download and install the script: cd /var/www/html wget http://citylan.dl.sourceforge.net/project/acrasteriskcdrr/rpt.tar.gz tar xzvf rpt.tar.gz then point your internet browser to http://your-ip-address/cdr.htm
