Open Source Mac VoIP Software - Page 5

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.

This project aims to provide Web Service clients wrapper for the Voxbone API (VoxAPI)

An XCAP server is used by XCAP clients to store data like buddy lists and presence policy in combination with a SIP Presence server that supports PUBLISH, SUBSCRIBE and NOTIFY methods to provide a complete SIP SIMPLE server solution.

Xen-Agi is an open source java library that allow you to build java applications that interact with the Asterisk Server for VOIP and PBX functionality. This project supports the FastAGI protocol exclusively and is inspired by the Asterisk-Java project.

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.

A simple lightweight instant messaging and VoIP client wirtten in Java. It implements the SIP protocol as described in RFC 3621 and other related RFCs and implementation requierements.

This program monitors a given email address for a certain subject with a list of phone numbers, and then calls them - all at once. Requires Skype.

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.

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

This is a SIP signaling layer to create a fully operative multipoint (video) conference server using SIP clients and RTP media streams in combination with strManager as a media management layer.

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 rules that adds top-up, purchase of add-on, assign/revoke permission, create and assign agent / extensions. ICTService also manages a form that allows new customers to choose a package to subscribe. Ratelist module directly connects with service portal's rate list and imports complete rate list in the Drupal table for rate list.

VoIP/SIP AutoDialer. Broadcasts a message with the option to transfer person to another number. Now includes new survey options.

Java SIP stack, originally based on mjsip.

miTester for SIP is an automated SIP testing tool designed and developed to take care of the complex pre-deployment testing of SIP applications easily. This SIP testing tool can be used to simulate SIP call-flows & automate functional, regression tests.

Administration Tool for Mumble -/- Murmur Servers. Mumb1e Admin Plugin, also named "MAP" is a powerful Murmur web interface application written with PHP that can create useres, edit themself and a lot more for Mumble server. MAP is developed over m

Nhabla-Bill: Is a report system a billing application inspired on Asterisk-Stat by Areski, offering many useful reports for the analysis, add to the billing function.

RUMPI: Real Time Multi Peer Interaction. RUMPI want to be as ubiquitous as Flash-browser-plugin, but enabling browsers for real time communication between peers: VoIP (IAX2) and Real-time video Streaming. RUMPI is not a plattform, is a Browser plugin,

This C++ library has been designed as a Chrome SIP stack. Sippet is an open-source SIP User-Agent library, compliant with the IETF RFC 3261 specification. It can be used as a building block for SIP client software for uses such as VoIP, IM, and many other real-time and P2P communication services. The main target was to enable Javascript applications to use UDP, TCP and TLS transports along WebSocket. Existing SIP solutions for the browser are forced to use the WebSockets API to send/receive SIP messages. But for many service providers, the WebSocket protocol could not be a feasible solution due to scalability, support and other business constraints. By the way, the stack can be used as a full client-side C++ Stack (full-featured) without Javascript integration.

skiputler makes your Topcom "Butler 4012 USB VoIP" phone work on Linux and Mac OS X. It's a userspace driver which connects the phone to Skype.

vTiger Freeswitch Integration by NYFON. Allows to perform outbound (Click to Call) and incoming (wip) calls from vTiger 6.1+ interface. Modified PBXManager allows to choose between Asterisk and Freeswitch for PBX integration. Same (and extended in the future) functionality as Asterisk interface.

The web application offers to import call protocols of phone calls and lists of billed calls from different telephone providers. They can be matched to existing call protocol entries to generate cross-provider phone bills.