Open Source PHP VoIP Software - Page 2
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PHP VoIP Software
View 383 business solutionsBrowse free open source PHP VoIP Software and projects below. Use the toggles on the left to filter open source PHP VoIP Software by OS, license, language, programming language, and project status.
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$300 Free Credits to Build on Google CloudPut your $300 in credit toward real workloads, then keep building with free monthly usage for 20+ products. No commitment and no charge until you upgrade.
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Fully Managed MySQL, PostgreSQL, and SQL ServerCloud SQL handles your database ops end to end, so you can focus on your app.
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Asterpoll is an addon to asterisk that allows a asterisk admin to create a poll question or questions . this poll is a destination on the PBX. callers answers are recorded and the results are shown in a graphical environment.
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Asterisk based callcenter system. Administration interface to manage queues, agents, accounts, skills - track calls, create statistics. Realtime queue monitor. Agent web interface with build in IAX softphone.
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DiWA (Dialplan Web Administrator) is a php/ajax interactive web application to manage Asterisk's diaplan. It supports both static and realtime configuration, and doesn't require database.
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Build Agents and Models on One PlatformGemini Enterprise Agent Platform is Google Cloud's comprehensive platform for developers to build, scale, govern, and optimize agents and models. Choose from Google's most advanced models and third-party models like Anthropic's Claude Model Family.
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FreePBX module for ENUM lookup API at enumplus.org.
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Open Source ITSP / Hosted PBX software that sits on top of Asterisk to handle customer management, rates via LCR, cdr reports, multiple phones per customer, customer portal, among many other things.
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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
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Enhanced Callback
Adds enhancements to Freepbx callback script
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Go from Code to Production URL in SecondsSkip the Kubernetes configs. Cloud Run handles HTTPS, scaling, and infrastructure automatically. Two million requests free per month.
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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.
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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
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FreeSentral is a full IP PBX consisting of a Linux Distribution, an IP PBX and a Web Graphical User Interface for easy configuration.
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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.
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A modularly designed system providing command/trigger functionality and response mechanism via SMS text messages
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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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Mongo has bundled Asterisk 1.4.13 + add-ons 1-4.4 + freePBX 2.3.1 + add-ons + sugarOS 4.5.1 + XAMPP 1.6.4 + MYSQL (with it's own sock) + PHP (part of XAMPP) + oscommerce
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MurMan, the Murmur Manager, is a user friendly, robust, and attractive web management interface for Murmur servers based on PHP, using Murmur's (server component for the Open-Source voice communication Mumble) Ice RCP.
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MumurWeb will be a comprehensive webinterface for Murmur servers based on PHP. Murmur is the server component for the Open-Source voice communication Mumble.
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Nhabla-Queue: Is a Asterisk report system for inbounds call center campaigns, provide many reports with useful information and graphics for analysis, which can be printed on PDF.
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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.
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This is a repository for QueueMetrics add-ons that are contribuited by the community and are open-source.
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OpenVOIP
Padrão de comunicação VOIP.
O OpenVOIP é uma solução Open Source para que empresas possam ter um padrão de comunicação VOIP. -
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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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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 scratch. PURPOSE: This project is designed to install the latest stable version of certified-asterisk-13.1-current + FreePBX V.12-based system in Debian 8.1 Platforms and versions tested: + 686 and amd64 + Debian 8.1 Jessie + Certified Asterisk 13.1-current - LTS + Libpri 1.4.4 + DAHDI COMPLETE LINUX Current + FreePBX 12. + Avantfax 3.3.3 + FOP2 2.29.02 + Webmin 1.760 + PHPSysInfo 3.2.2 + Fail2ban + CSF Firewall
