Showing 38 open source projects for "database"

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  • 1
    VoIP monitor

    VoIP monitor

    VoIP SIP and SKINNY quality analyzer and packet / audio recording tool

    ...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/RTCP/T.38/udptl protocols. VoIPmonitor can also decode audio.
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    Downloads: 612 This Week
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  • 2

    VOIP-VOICE-TO-TEXT&ANALYS

    Convert VoIP calls to text and analyze them with AI

    ...After each call, the audio file is sent to the GPT-4O AI engine, producing editable transcripts. The software also provides AI-powered call analysis, extracting key points, customer requests, satisfaction levels, and sensitive topics, all stored in the database. This helps sales and support teams make faster decisions, improve response quality, and enhance customer experience. Fully compatible with Issabel and open-source VoIP systems, the software runs securely on internal networks without external services. By combining speech recognition, intelligent analysis, and automatic call logging, it provides a professional and modern tool for managing organizational phone communications efficiently.
    Downloads: 0 This Week
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  • 3
    Elastix

    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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    Downloads: 178 This Week
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  • 4
    OpenSIPS/OpenSER-a versatile SIP Server
    OpenSIPS (former OpenSER) is an GPL implementation of a multi-functionality SIP Server that targets to deliver a high-level technical solution (performance, security and quality) to be used in professional SIP server platforms. IMPORTANT: this is no longer the main hosting for the project. This was moved on GITHUB - https://github.com/OpenSIPS/opensips
    Downloads: 21 This Week
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  • 5
    Accent Tool Suite

    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...
    Downloads: 0 This Week
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  • 6
    GNU Gatekeeper (GnuGk)

    GNU Gatekeeper (GnuGk)

    H.323 Gatekeeper for VoIP and videconferencing

    ...Please find current versions at https://www.gnugk.org/ The GNU Gatekeeper (GnuGk) is a full featured H.323 gatekeeper under GPL license. It supports VoIP and videoconferencing and includes Radius and database support and many call routing functions. The project has moved! Please find current versions at https://www.gnugk.org/
    Downloads: 9 This Week
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  • 7
    FreePBX (formerly Asterisk Management Portal) is a project to bring together best-of-breed applications to produce a standardized implementation of Asterisk complete with web-based administrative interface.
    Downloads: 6 This Week
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  • 8
    Passwords generator for IAD devices, using 3 keys and a index. The result is a fixed lenght of alphabetic characters. The algorithm works with other 3 big fixed prime numbers. These 3 numbers define a family of passwords. Finally, it provides a guaranteed functional string to use in all variety of devices without rejects.
    Downloads: 0 This Week
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  • 9
    cdr-view

    cdr-view

    Webface witch extract CDR (Call Detail Records) from MySQL base.

    Webface witch extract CDR (Call Detail Records ) directly from MySQL base. May be used as SOHO billing system.
    Downloads: 1 This Week
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  • 10

    cyber-wrt PBX

    Cyber-wrt pbx is embedded system in DD-Wrt router

    ############################################### #--- CYBER-WRT by Christian Zeler / Comdif Telecom ---# #----------------- http://cyberhotel.comdif.com ------------------# ############################################## You can find 2015 news Cyber-hotel versions on http://cyberhotel.comdif.com including cyber-hotel on freepbx - Cyber-hotel decktop. . Cyber-wrt is a complet Asterisk IPBX and Hotspot system embedded in DD-Wrt router. Last version is build for ASUS RT-AC56U...
    Downloads: 0 This Week
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  • 11
    Eternity smdr

    Eternity smdr

    perl based SMDR/CDR capture & display on Web based PHP with mysql DB

    You can Capture Matrix Comsec' Eternity/NAVAN product Online SMDR/CDR using Perl Script. Store it in MYSQL Database and display it by using filter in PHP
    Downloads: 0 This Week
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  • 12
    OSDial

    OSDial

    OpenSource Call Center and Dialer System

    An Open Source (predictive) Dialer. OSDial is a full featured GPL dialer which does Manual, Power and Predictive dialing out of the box. Its easily installed through one of our .iso images. Install manual available at http://store.osdial.com/ Full details available at web site (http://osdial.com).
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    Downloads: 12 This Week
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  • 13

    Fix Skype

    Fix Skype fixes future timestamps in the Skype messages database.

    Fix Skype is a python script and Win32 binary that is used to fix "future" timestamps in the Skype message database, such as those caused by the end of DST or by dual-booting a Linux system with time set to UTC from anywhere west of GMT. Essentially, you feed it your username and the number of hours by which some timestamps are in the future, and it subtracts that amount of time from any timestamp greater than "now".
    Downloads: 0 This Week
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  • 14
    KAMAILIO (OpenSER) - robust, secure and scalable Open Source (GPL) SIP (RFC3261) server implementation with large features set (over 90 extension modules). As of May 2009, source code is hosted by GIT repository at http://sip-router.org
    Downloads: 1 This Week
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  • 15
    VoIPGMap

    VoIPGMap

    Graph active VoIP calls on Google Maps.

    A bash shell script which will read active call destinations in E.164 format from a source (such as Asterisk), then run the called number through some plain text and MySQL databases to find the best latitude/longitude match. It then creates a .html file with Google Maps API JavaScript code which will display a single call origination point and the active destinations (calls) on a Google Maps world map.
    Downloads: 0 This Week
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  • 16
    An open source software project delivering converged voice and data applications. The StarPound platform enables you to rapidly develop voice and data solutions using the power of business process modeling. More info: http://www.starpound.org.
    Downloads: 1 This Week
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  • 17
    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
    Downloads: 0 This Week
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  • 18
    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.
    Downloads: 0 This Week
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  • 19
    ENUMPlus
    FreePBX module for ENUM lookup API at enumplus.org.
    Downloads: 0 This Week
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  • 20
    Open UC Framework (openucf) is a unified communication server which supports telephony, presence, instant messaging and contact management.
    Downloads: 0 This Week
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  • 21
    SkypeXMPP is a Karaka™ fork aimed to provide Skype to XMPP gateway It supports presence, instant messaging, multi-user chat and voice (thanks to Freeswitch) This product uses the Skype™ API but is not endorsed or certified by Skype.
    Downloads: 0 This Week
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  • 22
    Quimerus PAR-CENT
    [es ] Aplicación web central para particulares. [en ] Core web application for individuals. [fr] Application Web centrale pour les personnes. [pt] Central de aplicação web para indivíduos. [ar] المركزي تطبيق ويب للأفراد
    Downloads: 0 This Week
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  • 23

    OpenBTS

    GSM L1-L3 stack with SIP network interfaces

    OpenBTS is an implementation of the GSM air interface (Um) that allows cellular handsets to be used directly as SIP endpoints. It uses a software-defined radio to generate its air interface and uses Asterisk or yate as its network interface.
    Downloads: 0 This Week
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  • 24

    astrtr

    Asterisk module to route traffic (very obsolete)

    astrtr is a FreePBX module for Asterisk to route calls from one trunk to another. Makes Asterisk PBX a VoIP Switch as well. Includes multiple routing tables, max connections per trunks, some reports, CDRs, etc.
    Downloads: 0 This Week
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  • 25

    SerMyAdmin

    Web interface for OpenSIPS

    I'm sad to say that, unfortunately, the serMyAdmin project is being discontinued starting in September 1st, 2010. After three years of development, the project no longer makes sense. The main reason to create the serMyAdmin was the lack of a GUI for OpenSER/OpenSIPS. The OpenSIPS control panel fulfill this need and it is not wise to maintain two GUI projects. Our team will support the opensips-cp initiative to avoid splitting the community between two open source projects for OpenSIPS GUI. I...
    Downloads: 0 This Week
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