Search Results for "sofia-sip"
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Showing 26 open source projects for "sofia-sip"
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HubSpot is an AI-powered customer platform with all the software, integrations, and resources you need to connect your marketing, sales, and customer service. HubSpot's connected platform enables you to grow your business faster by focusing on what matters most: your customers.
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Stop messing with tools that aren’t designed to amplify volunteer programs. With VolunteerMatters, it’s a delight to manage everything in one place.
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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... -
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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... -
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Cisco TMS PhoneBook - PHP Version
Emulates Cisco TMS PhoneBook for CMM or Endpoints with PHP
... 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. -
PRTG Network Monitor is an all-inclusive monitoring software solution developed by Paessler. Equipped with an easy-to-use, intuitive interface with a cutting-edge monitoring engine, PRTG Network Monitor optimizes connections and workloads as well as reduces operational costs by avoiding outages while saving time and controlling service level agreements (SLAs). The solution is packed with specialized monitoring features that include flexible alerting, cluster failover solution, distributed monitoring, in-depth reporting, maps and dashboards, and more.
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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)
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... -
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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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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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Textellent's business texting solution makes designing, managing, measuring, and integrating SMS and MMS campaigns easy. Whether used for customer service, sales, or marketing, Textellent supports your customer journey with an easy-to-use service that text-enables local business lines and serves businesses of any size. Robust scheduling and appointment reminders are also available, as are keywords and shortcodes for easy opt-in programs with TCPA compliance supported by AI.
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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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VitoshaTrade is Distributed Artificial Neural Network trained by Differential Evolution for prediction of Forex. Project development is in Sofia, Bulgaria. Vitosha is a mountain massif, on the outskirts of Sofia, the capital of Bulgaria.
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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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Eternity 2 puzzle heuristic solver is distributed computing project trying to solve 2 million dollars puzzle by using heuristic algorithms. Project development is based in Sofia, Bulgaria. Eternity 2 website: http://uk.eternityii.com/
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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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vTiger - FreeSWITCH Integration
Extend vTiger CRM to add FreeSWITCH support
Current vTiger CRM integration is to control the Asterisk (soft) PBX to establish and log calls. 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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ADoWI is a web interface with management of Asterisk SIP accounts and phone book for the moment. ADoWI is in french : "Asterisk Directory on Web Interface". It don't need a new installation of Asterisk.