Search Results for "voip php" - Page 3

A simple interface to work with SIP numbers of Asterisk. Add, delete, and edit SIP numbers in the MySQL database As well as viewing CDR records Простой интерфейс для работы с SIP номерами сервера Asterisk. Добавление, удаление и редактирование SIP номеров в базе MySQL А так же просмотр CDR записей

Interpedia is moved to https://bitbucket.org/interpedia/interpedia/downloads --------------------------------------------- Interpedia VLMS is a cloud based (PaaS,SaaS) product: - for the administration - documentation - tracking - collaboration - and reporting of training programs - virtual classrooms - online events - e-learning programs - and training content - Learning Management System (openeclass) It provides advanced functionality by supplying fully functional...

The Agispeedy is an AGI Application Server implemention in asterisk and written by Pure PHP. it inconceivable faster than normal agi about 1000%, and improves performance.

This is a vTiger module adding predictive dialing features using the popular asterisk PBX. The module calls the leads included in active campaigns tuning the number of concurrent calls according with the number of available agents, the answer ratio and other parameters (dial time, conversation time etc.) It is included also an optional module providing statistics on the contact center behaviour

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.

The goal of Xplico is to extract the applications data from an Internet traffic capture. For example, from a pcap file Xplico extracts each email (POP, IMAP, and SMTP protocols), all HTTP contents, each VoIP call (SIP), and so on. NFAT

Elastix SMS Module is a module for sending individual and bulk SMS messages. You can create any number of SMPP trunks, create groups of numbers, create SMS campaigns and send individual SMS messages. More info about us in: http://www.iberoxarxa.es http://www.facebook.com/iberoxarxa

OpenCallshop for VoIP Switch was started to be an alternitive to VoipSwitch Inc. closed source callshop systems, You can extend development of this project and customize it to fit your needs. if you would like to get a custom developed callshop desktop or web software do not hesitate to contact me. if you're reseller this product consider a donation.

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.

GoAutoDial is an all-in-one open source Vicidial administrator utility that enables you to manage your Vicidial - CentOS based server. Package includes Vicidial web admin, GoAutoDial admin, NEW AgentUI. For more info, please visit www.goautodial.com.

Ex Elaxtix community and others groups , migrating web panel for Debian base distros

The Expresso Livre is a collaborative software that gathers functions of electronic mail, calendar, contacts center, workflow, filemanager and messenger developed starting from Egroupware.Several improvements were developed including easier interface

FreePBX module for ENUM lookup API at enumplus.org.

FreeSentral is a full IP PBX consisting of a Linux Distribution, an IP PBX and a Web Graphical User Interface for easy configuration.

This is the official location of the Endpoint Manager Module for FreePBX

Open UC Framework (openucf) is a unified communication server which supports telephony, presence, instant messaging and contact management.

Software designed to rapidly contact many individuals through Phone (Asterisk), SMS, Email and a VHF Paging system. The designed application is for emergency response agencies requiring an inexpensive way to manage their membership and deployment.

Open source software components of the AastraLink Pro 160 Supplied from CDROM "71-001049-02 REV01" by Aastra Telecom

Fluent is a suite of tools for the management of diverse VoIP networks. Fluent RTE, a reporting and templating engine, is at the core. Add-on modules are available to manage provisioning, CDR, feature sets, routing and more.

Live-cd com Servidor SIP baseado em opensips. O Servidor de Telefone Livre transformará um simples PC numa central SIP multidominio. Estamos construindo a documentacao do projeto na wiki do sistema trac.

It is an appliance software that integrates the best tools available for Asterisk-based PBXs into a single, easy-to-use interface. include astercrm astercc voicemail,fax,et

The web application offers to import call protocols of phone calls and lists of billed calls from different telephone providers. They can be matched to existing call protocol entries to generate cross-provider phone bills.

A modularly designed system providing command/trigger functionality and response mechanism via SMS text messages

trixbox CE is an easy to install, VOIP phone system based on the Asterisk PBX. trixbox is designed for home or office use. trixbox CE includes CentOS linux, mysql, and all the tools needed to run a business quality phone system. (formerly asterisk@home)

PHP based Viewer for Voice Servers like Mumble.