Search Results for "alpha"
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Showing 45 open source projects for "alpha"
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SIPSL (SIP Service Layer) is a Programmable Session Border Controller (SBC) and SIP B2BUA, as defined in RFC3261. Written in c++ and multithreaded. The Application logic can be implemented by extending the application class and implementing the call backs.
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asterisk-kannel
Asterisk and kannel integration
This project aims to integrate Asterisk IP-PBX with Kannel SMS gateway. -
Stop Cyber Threats with VM-Series Next-Gen Firewall on AzureGain integrated visibility across all traffic in a single pass. Deploy Palo Alto Networks VM-Series to determine application identity and content while automating security policy updates via rich APIs.
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VoIP Honey project provides a set of tools for building an entire honeynet, thus includes honeywall and honeypot emulating VoIP environments such as Asterisk PBX or OpenSer with fully configurable connections. Voip Honey runs on GNU/Linux and Windows Systems. It can be compiled for Mac OSX as well.
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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. -
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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 -
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SIP Anonymization Tool (SiAnTo)
Small and effective program for SIP traces anonymization
The Session Initiation Protocol (SIP) is a signaling communications protocol widely used nowadays for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks. A good way to design optimization techniques for SIP deployment would be to analyze SIP traffic from existing networks. However, publicly available analyses of SIP traffic are rare and thus not a lot of knowledge exists about typical behavior of a SIP server (as opposed to, for... -
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ictbilling
A client management, billing and e-commerce solution
ICTBilling contains a set of Drupal 7 modules that can be used with Ubercart E-commerce solution and ICTBroadcast, a unified communication auto dialer software. This Client and Billing solution for ITSP has following modules and rules: client module, ictservice module, ratelist module and rules for ubercart. Client module handles clients user interfaces, subscribed service details, service links, list of available addons and current balance. Another module, ICTService creates simple... -
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cdr2serial
Send Asterisk CDR to serial device
cdr2serial sends Asterisk CDR serial port for the use of software for hotels. They have been used libraries asterisk-java for agi events and jssc for send via serial port. -
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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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MobilIVR is essentially a convenience layer that sits on-top of Asterisk [www.asterisk.org], as well as Kannel (SMS gateway).
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Hanasu
A P2P Messenger which uses JXTA. It provides secure RSA/AES end-to-end
Does not work because there is no master server! A P2P Messenger which uses JXTA. It provides secure RSA/AES end-to-end encryption and VoIP-Capabilities. -
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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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agispeedy
The Agispeedy is an AGI Application Server implemention in asterisk
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. -
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vtdialer
predictive dialing for vTiger using asterisk
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 -
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Zanzibar is a complete, standards based IVR. It includes an MRCPv2 Server with ASR and TTS engines as well as an voiceXML interpreter so that you can deploy and run voiceXML applications. It integrates with VOIP PBX’s (like Asterisk) using SIP and RTP.
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node (nodejs) non-blocking RADIUS client wrapper
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This is a open source media gateway controller. The protocol used is MEGACO/H.248. In a later version also SIP will be supported.
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A $10 Analog Telephone Adapter (ATA) that allows low cost routers such as the WRT54G to connect to an analog telephone. The application is low cost VOIP, in particular for the developing world.
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The goal of the project is to create a high-performance, open-source and standards-compliant implementation of a Home-Subscriber-Server (HSS) for use in a IMS context.
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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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SIP.NET is SIP stack .NET library written in C#. SIP message parser realized on Deterministic Finite-state Machine (DFA parse all SIP message for one pass). DFA generates automatically from Augmented Backus–Naur Form from RFC3261.
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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. -
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The project "Sippie" is an additional tool for automatically converting Wireshark-Traces to XML - scenarios, which can be easily used as infile for the OpenSource SIPp test tool and traffic generator.
