Open Source Mac VoIP Software - Page 4

This project serves to provide respins of some common open source PBX projects to include additional drivers, functionality, and software not found in the original projects.

Spice up your Mumble experience using this recreation of 4 different skins merged and adjusted into a new one. It is tested and running stable using Mumble 1.2.x. To install: Download, extract and follow the readme.txt contained in the archive. All credits go to the devs of the individual pieces including Mumble itself.

SIP Inspector is a tool written in JAVA to simulate different SIP messages and scenarios. You can create your own SIP signaling scenarios, customize SIP messages and monitor incoming and outgoing messages. The tool can play RTP streams from a pcap fi

Project moved to GitHub. https://github.com/davidfischer-ch/smpte2022lib

This is selective answering script for Asterisk PBX.

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 would like to say thanks to all the people that in some way contributed with the project. The svn code is available to anyone who wants to maintain the project. Most of our development efforts are now focused in a new commercial product based on OpenSIPS called SIP Pulse, completely rewritten in Java and Glassfish www.sippulse.com supporting pre and post paid users with billing and reseller portal.

An adhoc network simulator that tests multiple transceiver base stations in a peer to peer environment.

A Sip live audio feeding agent. The agent captures audio from sound card and sends live audio stream(uLaw) to caller(sip phone) using RTP. It is based on Peers 0.3(http://peers.sourceforge.net/). Can be used in IP telephony to broadcast live audio.

This project provides a meter of QoS on VoIP using the protocol SIP Media Loopback

Open Source implementation of SIP stack according to RFC 3261 and related standards.

SIP to Skype Gateway/Bridge/Converter/Adapter. Make and receive SIP to Skype Calls and Skype to SIP Calls. Call Skype users using speed dial or use SkypeOut. Make SIP calls from Skype using a SIP provider or SIP PBX. Use as a Skype Trunk with a PBX.

jSipgateConnect library allows Java-based access to all functions specified in the 'Sip Advanced Messaging, Usability, Routing and Accounting Interface' specification.

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.

This project has been relocated to http://github.com/sippy/rtpproxy http://github.com/sippy/b2bua

A simple status bar item that sets your Skype mood message to the song that is playing on iTunes, VLC or Spotify.

THUNDER!Box synchronisiert das Thunderbird-Adressbuch an einen FRITZ!Box DSL-Modem/Router. --- THUNDER!Box synchronizes the Thunderbird addressbook to the FRITZ!Box DSL-Modem/Router.

It's a teamspeak viewer based on "Teamspeak Viewer", but made in AJAX the XML is generated with PHP the display is done with javascript

openssl is well-known allover the world,but it is huge,complex.our target is develop a tinytls,it is small,simple,good understanding,and usually for embedded system.But it can support most openssl features.You can add some other features easily.

Open source callcenter suite focused on high performance and easy administration.

Turn.Message is .NET parser / generator of the TURN (Traversal Using Relay NAT) messages written in C#. Compatible with [MS-TURN] extension and clients based on Microsoft UCCAPI SDK.

Twinkle is a VOIP based telephony software on linux. This project will install this software on MAC OS X using fink package management system. This will be included in stable tree of fink.

The VoIP voice-to-text software for Issabel is an intelligent, AI-based solution that converts calls into accurate Persian text. 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.

This is an initiative to assemble/develop the cheapest, easiest to setup, easiest to manage, scalable, Open Source, standards-based, wireless local do-it-yourself telephone company toolkit in the world.

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 telephone system for your company, save, take advantage of recent innovations and provide opportunities to integrate your business processes if you wish to do so.

Visual voicemail viewer for Asterisk/AsteriskNOW written in PHP. Users log in with their extension and v/m password and can download messages with the web browser.