Open Source Windows VoIP Software - Page 5

Plug and play provisioning server for snom VoIP phones written in Python. The server supports the plug and play provisioning feature embedded in snom phones of the snom 3xx, snom8xx and snom MeetingPoint series.

Application to remote control Polycom VVX IP phones via their web interface (using HTTPS). This application is initially intended to perform certain actions on phones running in Microsoft Lync mode. These actions include: * get device information * get status * sign in using PIN authentication * sign out * reboot * factory reset It also supports performing actions in batch reading data from a CSV file.

Pops up a URL when a Call on an Asterisk Server is Answered, passing the Extension, Caller ID name and number. Connects to the Asterisk Server using the Manager Interface. Can be used to connect to a web based CRM.

VOIP client/server in python >= 2.6. Audio in/out: ossaudiodev (UNIX like) or SoX. Network: bzip2 compression, speex or ogg audio compression, you can configure all, minimum bytes per second: 350-400 in speex mode: U8, 6 kHz, quality 0, bzip2, buf 4K

QoS Number is a project to control de quality of service ( QoS ) in a IP-Centrex Plattform .

[es ] Aplicación web central para particulares. [en ] Core web application for individuals. [fr] Application Web centrale pour les personnes. [pt] Central de aplicação web para indivíduos. [ar] المركزي تطبيق ويب للأفراد

Adds a Right Click context menu to your browser to call number from Lync. Highlight a piece of text in IE/FF13/Chrome and the script will strip out non numerics and pass to Lync via tel: handler and Lync can normalise the number.

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.

IM,WebIM and Sip voip tool. Client is Win32, Server is CentOS 5. support video,audio, file trans, sip call.

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

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.

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.

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.

SipTunnel is used to tunnel SIP UDP datagrams (and also RTP datagrams) through NAT using TCP protocol. It also can be used for many other purposes.

Sipflanker helps you find IP phones and IP PBXs in your network that may have Web GUIs available for browsing. It also shows their default usernames and passwords so you can test them and make sure they are not vulnerable to attack.

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

PLEASE NOTE So there is no way of making the current Skype accept the .lang file. You cannot run a version older than Skype 7 and even installing 7-8 is fiendishly complicated because it forces everyone into the app unless you run it in compatibility mode. Time to call it a day. I will back up the current files and then shut down the project at the end of the month. Please make your own backups from SourceForge because I till take that down too. If anyone wishes to take it on to handle just the Linux part, please give me a shout. Thank you for all your hard work over the years and really sorry to see this go but it's probably time to find an alternative to Skype that's localizable and OS.

Spark plugin for integratión with CentricCRM. This plugin for Spark XMPP client complements the integration between Asterisk PBX, Centric CRM and XMPP Server. Automatically open session on Centric CRM and redirect to contact the page