Search Results for "sip"
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Showing 24 open source projects for "sip"
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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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Demo Series - Small Business Backup By VeeamWatch this on-demand demo series and learn how to protect your Microsoft 365 data with clear, simple, actionable steps that are easy to implement for businesses of all sizes.
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VoIP monitor
VoIP SIP and SKINNY quality analyzer and packet / audio recording tool
VoIPmonitor is open source network packet sniffer with commercial frontend for SIP SKINNY MGCP RTP and RTCP VoIP protocols running on linux. 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. ... -
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ictcore
Unified communications framework and REST API for developers
ICTCore is the communications framework the rest of our open source stack is built on. It sits above FreeSWITCH and gives web developers a REST API for voice calls, fax, SMS and email instead of a dialplan to learn. The idea is that a communications workflow should be something you write in your own application. You describe what should happen, and ICTCore deals with the switch, the media and the state machine underneath. It is the engine behind ICTFax and the open source ICTDialer, so... -
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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... -
$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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Cisco TMS PhoneBook - PHP Version
Emulates Cisco TMS PhoneBook for CMM or Endpoints with PHP
...Anyway, this tool emulates the TMS PhoneBook and allows you to search and add new participants to a existing conference. The search is regular expression and the filter compare the user query with the endpoint name or H323/SIP dial string from the database. All you need is a Web server (tested with apache) with PHP and "RewriteEngine on" to make it work Optionally you can use a SQL server to have a real database, but I used a simple array as db. -
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TeeBX
Asterisk based VoIP pbx appliance
TeeBX wants to be a VoIP appliance with a user interface to easy manage a communication platform based on Asterisk (forked from askozia pbx). Website, support, documentation (coming soon): http://www.teebx.com/ -
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Xplico
Xplico is a Network Forensic Analysis Tool (NFAT)
...The goal of Xplico is extract from an internet traffic capture the applications data contained. For example, from a pcap file Xplico extracts each email (POP, IMAP, and SMTP protocols), all HTTP contents, each VoIP call (SIP, MGCP, MEGACO, RTP), IRC, WhatsApp... Xplico is able to classify more than 140 (application) protocols. Xplico cam be used as sniffer-decoder if used in "live mode" or in conjunction with netsniff-ng. Xplico is used also in CapAnalysis: http://www.capanalysis.net -
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TOFFEE-DataCenter
Linux Server and Data-Center WAN Optimization
TOFFEE-DataCenter is the new TOFFEE variant meant for Data-Center, Server, HPC, Load Balancing, Cluster/Cloud Computing and SDN deployments. With TOFFEE-DataCenter (and or TOFFEE), OEMs and other equipment vendors can make use of this platform/stack, integrate, port and build WAN Optimization devices/appliances on to their existing product range. Or build a dedicated WAN Optimization hardware product series via the same. -
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Asterweb is an Asterisk Realtime Configuration utility written in PHP. It configures the realtime settings for voicemail, extensions and sip buddies. Users can login with their voicemail user and pin and check their voicemail.
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Train ML Models With SQL You Already KnowBuild and deploy ML models using familiar SQL. Automate data prep with built-in Gemini. Query 1 TB and store 10 GB free monthly.
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Git repo: https://github.com/asipto/siremis Web management interface for Kamailio (OpenSER) - handle subscriber profiles, access control lists, accounting, least cost routing and load balancing, monitoring charts, xmlrpc communication with SIP server
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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
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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
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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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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.
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Asterisk Realtime Administration. Administer realtime extensions, SIP users, voicemail all in realtime via a web interface. Works with the mysql asterisk realtime extension.
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Open Source IM and voice client using Jabber and SIP protocols, with great audio quality thanks to speex and a nice and clean interface. Programmed in Java and C++. It supports chat, voice and file transfers.
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IM,WebIM and Sip voip tool. Client is Win32, Server is CentOS 5. support video,audio, file trans, sip call.
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WIST is a PHP Web Interface who permits you to connect on a remote host/port and listen/filter a SIP dialog of an specific SIP From number.
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Distributed SIP Analyzer is a SIP protocol analyzer for Unix. It allows you to examine SIP from different local area network. You can interactively browse the capture data, viewing callflow sequence diagram and detail information for each SIP Session.
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Very simple and easy to use self-hosted blog system like wordpress, but sip is most interactive and easy to use whit many new feautures. It's written in PHP and using MySQL database.
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uPBX aim is to give the user a GUI to manage all the interesting features available on the CallWeaver PBX software, such as T38, SIP, H323, PSTN, Conferences, queues and many more.
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vTiger - FreeSWITCH Integration
Extend vTiger CRM to add FreeSWITCH support
...This call is established by the vTiger user of the vTiger web based GUI. FreeSWITCH can be configured as a similar soft PBX and so the goal is to get vTiger to cause Freeswitch to establish SIP connections from the vTiger user's phone to the customer phone.
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