Search Results for "using"

The NCID (Network Caller ID) project is Caller ID (CID) distributed over a network. The project contains the NCID package and 4 optional client packages. Each package is described at the NCID web site. A non-inclusive list of 3rd party addons is also available at the web site Available Packages: NCID - contains the server, gateways, and a client with output modules LCDncid - a client that uses LCDproc to display Caller ID on a LCD display NCIDandroid - a client and gateway for...

Note: for binaries and installation instruction please visit official website. http://ictfax.org/ ICTFax is Open Source Fax Server software to send and receive Fax over Internet & Emails developed over Freeswitch, ICTCore and Angular Framework , It works with T.38 and G.711 pass through and feature email to fax, fax to email, web to fax, fax to web and also support fax machines to send and recieve Fax using ATA. Installation Guide https://www.ictfax.org/ictfax-installation-guide/ Administration Guide https://www.ictfax.org/ictfax-admin-guide/ User Guide https://www.ictfax.org/ictfax-user-guide/ Commercial Support If you do not posses Linux / VoIP expertise to install and setup ICTFax or you are busy, We can help you in installation, configuration and testing to make sure ICTFax setup is functional and tested. ...

Multi user address book application accessable through the web. Store, edit, search and export addresses, phone numbers, … using a web browser. Code moved to https://bitbucket.org/icemac/icemac.addressbook Documentation see https://icemacaddressbook.readthedocs.io/en/latest/ New releases (after 6.0.2) see https://pypi.org/project/icemac.addressbook/#history

The BOT is used in our Discord server.

...It simplifies service creation to drawing diagrams, clicking buttons and defining a few tool parameters. All key functions of Cress can be controlled from the Chive diagram editor. Cress is written using portable and widely used languages (Perl and Java). It can be extended by a developer for other kinds of services that can be characterised by activity flows.

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.

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. Aim of this project is to offer a simple and effective, yet well-configurable, tool allowing for SIP trace filtering - the SIP Data Filter (SiDaFir).

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.

Twisted Protocols for communication with FreeSWITCH PySWITCH is a Twisted Python library that allows you to communicate with FreeSWITCH using inbound and outbound EventSocket connections.

...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

One of the first secure telephony programs. Not limited to VOIP but can also directly encrypt a telephone line by using modems. Calls are encrypted using Diffie-Hellmann key exchange and several symmetric ciphers. Download is possible from Germany only, see homepage

Turn.Message is .NET parser / generator of the TURN (Traversal Using Relay NAT) messages written in C#. Compatible with [MS-TURN] extension and clients based on Microsoft UCCAPI SDK.

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.

This project aim to provide a web based application to plan Conference Call using Asterisk application MeetMe(). The engine is based on Vaadin language to improve the portability of the application, Hibernate and some AGI script based on PHP.

An open source CTI client for Asterisk using the Asterisk Manager API. Uses pygtk and pyAsterisk.

SafiServer is an engine that powers the applications (called Saflets) created using SafiWorkshop. SafiServer uses these applications to control communications (i.e. Asterisk) along with a multitude of other applications (database, email, etc).

Java API to send SMS and WAP Push messages using Kannel SMS/WAP Gateway

Quintum Tenor VoIP Gateway Call Detail Record (CDR) Advanced Server. This CDR Server can collect Call Detail Records (CDRs) from multiple Tenors. The data is collected simultaneously and continuously. Supported MySQL and file storages.

This is a SIP signaling layer to create a fully operative multipoint (video) conference server using SIP clients and RTP media streams in combination with strManager as a media management layer.

Experimental open source project, which aims to test possibility of VoIP calls "from browser to browser". Main part is Java application implementing UDP hole punching, which is sent to both clients using JWS. See it online in action at project homepage

Twinkle is a VOIP based telephony software on linux. This project will install this software on MAC OS X using fink package management system. This will be included in stable tree of fink.

A Sip live audio feeding agent. The agent captures audio from sound card and sends live audio stream(uLaw) to caller(sip phone) using RTP. It is based on Peers 0.3(http://peers.sourceforge.net/). Can be used in IP telephony to broadcast live audio.

Using the new C# Dll from swig portability of Openobex/ObexFtp to write a nice GUI able to transfer file between two devices with obex protocol. More Info’s here: http://cs-obexftp.wiki.sourceforge.net/

PHP based Web-App amied at managing call-centers Outbound campaigns. Customizable call results and phone interview using external XML file.

FATS - FATS is a Twisted and Fast Asterisk's Telephony Services. Project contains implementation of FastAGI, AMI protocols for the Twisted framework. Using it you can develop fast and pretty services for the Asterisk IP-PBX.