voip conferencing
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Open Conference Systems 2.0
Open Conference Systems is a free Web publishing tool that will create a complete Web presence for your scholarly conference. more>>
Open Conference Systems or OCS for short, is a free Web publishing tool that will create a complete Web presence for your scholarly conference.
Main features:
- create a conference Web site
- compose and send a call for papers
- electronically accept paper and abstract submissions
- allow paper submitters to edit their work
- post conference proceedings and papers in a searchable format
- post, if you wish, the original data sets
- register participants
- integrate post-conference online discussions
Enhancements:
- manage conferences that occur more than once (e.g. yearly)
- an expanded, multiple-round review system
- e-mail template system
- localization & translation tools
- credit card payment for registrations
- automated "thanks-for-submitting" messages
- accept login/password for participants
- Creative Commons licensing of presentations
- more customizable, scalable and secure code
- ...and much more.
<<lessMain features:
- create a conference Web site
- compose and send a call for papers
- electronically accept paper and abstract submissions
- allow paper submitters to edit their work
- post conference proceedings and papers in a searchable format
- post, if you wish, the original data sets
- register participants
- integrate post-conference online discussions
Enhancements:
- manage conferences that occur more than once (e.g. yearly)
- an expanded, multiple-round review system
- e-mail template system
- localization & translation tools
- credit card payment for registrations
- automated "thanks-for-submitting" messages
- accept login/password for participants
- Creative Commons licensing of presentations
- more customizable, scalable and secure code
- ...and much more.
Download (1.5MB)
Added: 2007-06-16 License: GPL (GNU General Public License) Price:
864 downloads
VoiceOne 0.5.0
VoiceOne project is an easy to use web based GUI for the Asterisk PBX. more>>
VoiceOne project is an easy to use web based GUI for the Asterisk PBX. It gives full control over PBX settings from the admin configuration panel.
You can manage extensions, VoIP trunks, users queues and rules sets, and dynamically create a IVR.
Main features:
- Client/Server architecture based on Web services
- Relies on Asterisk Realtime Architecture (ARA)
- SIP extensions management (support for Zap/IAX/mISDN soon added)
- Remote offices via IAX with RSA public key encryption
- Supports VoIP and traditional Telco providers
- Powerful IVR creation system
- Queues system management
- Customizable users profile
- Powerful configuration of mISDN ports/interfaces (thanks to guys at beroNet for their support)
- Policies system for users/groups management
- SIP, IAX and mISDN general/default options configuration
- Static-like text editor for conf files
- Easy setup wizard
- lots more...
Enhancements:
- The IVR implementation is now completely dynamic and database based.
<<lessYou can manage extensions, VoIP trunks, users queues and rules sets, and dynamically create a IVR.
Main features:
- Client/Server architecture based on Web services
- Relies on Asterisk Realtime Architecture (ARA)
- SIP extensions management (support for Zap/IAX/mISDN soon added)
- Remote offices via IAX with RSA public key encryption
- Supports VoIP and traditional Telco providers
- Powerful IVR creation system
- Queues system management
- Customizable users profile
- Powerful configuration of mISDN ports/interfaces (thanks to guys at beroNet for their support)
- Policies system for users/groups management
- SIP, IAX and mISDN general/default options configuration
- Static-like text editor for conf files
- Easy setup wizard
- lots more...
Enhancements:
- The IVR implementation is now completely dynamic and database based.
Download (2.2MB)
Added: 2007-07-09 License: LGPL (GNU Lesser General Public License) Price:
843 downloads
Minisip VoIP Provider
The routing in incoming calls(answering machine, transmission, for example) is configurable in the webinterface. more>> <<less
Download (4.19MB)
Added: 2009-04-26 License: Freeware Price: $0
198 downloads
Konference 0.1 Alpha 2
Konference is a Video-Conferencing app for kde3. more>>
Konference is (or better: will be) a video-conferencing application for KDE. Since the rewrite (2005/01/25) it supports SIP as the signalling protocol. No longer H323 folks.
Enhancements:
- Added support for transmitting an image instead of video, added support for Arts, fixed various segfaults and reworked configure-dialogs a bit
<<lessEnhancements:
- Added support for transmitting an image instead of video, added support for Arts, fixed various segfaults and reworked configure-dialogs a bit
Download (1.2MB)
Added: 2005-06-16 License: GPL (GNU General Public License) Price:
1590 downloads
iConfKit for Linux 1.0.6
It is a free, easy-to-use, flash based lightweight video conferencing server. more>> ICONFKIT is a free, easy-to-use, flash based lightweight video conferencing server and messenging system built for small teleworking, collaborations and family contacts, with it you can contact your friends and colleagues, record your video and sound by webcam and microphone into files at server side and then play them later, make video conferences and textual group messenging, you can also share your ideas with some others via a whiteboard(doodle)!
Features:
* Zero install at client side
User interface at client side is a flash swf file, so that end users of this conferencing toolkits need not install anything except for a flash enabled web browsers, for instance Internet Explorer 6 or 7, Mozilla Firefox, that are usually preinstalled within every mainstream operating system. Your people can access the server at office from around the world to talk and collaborate.
* Platform independent
Currently iconfkit server can be deployed on various operating systems, including windows(i386) and linux(i386). other operating systems support will keep coming. Installers for x86_64 architecture has been scheduled.
Thanks to the adobes flash player distribution, the client can also run on many platforms(windows, linux, macosx, solaris, ...) without installing a bit of extra things.
* One small sized executable
The server programs size is very small, on most platforms the executables file size is no larger than one mega bytes. They can be easily put within portable storages.
Additional DLLs or other files are not required. The single executable provides all the functions needed by the system to run.<<less
Download (582KB)
Added: 2009-04-28 License: Freeware Price:
178 downloads
VoiceBuntu 1.02
VoiceBuntu (Ubunterisk) is an Ubuntu based live Linux distribution that uses Asterisk and VoiceOne to provide VoIP service. more>>
VoiceBuntu (Ubunterisk) is an Ubuntu based live Linux distribution that uses Asterisk and VoiceOne to provide VoIP service.
VoiceBuntus focus is to run asterisk with no installation needed. VoiceBuntu - ubuntu itself - has built in boot prompt cheap code one can use called persistent.
This feature allows the user to use VoiceBuntus persistent feature in order to keep settings on a flash disk or memory stick.
To start, just burn the ISO file on a CD-ROM then start the live cd distribution and wait until everything is ready.
Please note, that apache will start at the very last moment of the boot cycle. So you may a to wait 1 or 2 minutes to let everything load.
Once everything is prepared, start your browser and point to the IP address of your running VoiceBuntu (i.e. http://192.168.0.10).
If you have no DHCP Server available you can press F6 at boot screen and add additionally add the command parameter
static=xxx.xxx.xxx.xxx
netmask=xxx.xxx.xxx.xxx
bcast=xxx.xxx.xxx.xxx
gateway=xxx.xxx.xxx.xxx
dns1=xxx.xxx.xxx.xxx
dns2=xxx.xxx.xxx.xxx
where xxx.xxx.xxx.xxx is the IP of your choice.
By that VoiceBuntu boots with static IP address on eth0 network device.
Enhancements:
- This live CD release is based on Ubuntu 7.04 and VoiceOne 0.5.0_2 due to a minor bug in VoiceOne version 0.5.0.
<<lessVoiceBuntus focus is to run asterisk with no installation needed. VoiceBuntu - ubuntu itself - has built in boot prompt cheap code one can use called persistent.
This feature allows the user to use VoiceBuntus persistent feature in order to keep settings on a flash disk or memory stick.
To start, just burn the ISO file on a CD-ROM then start the live cd distribution and wait until everything is ready.
Please note, that apache will start at the very last moment of the boot cycle. So you may a to wait 1 or 2 minutes to let everything load.
Once everything is prepared, start your browser and point to the IP address of your running VoiceBuntu (i.e. http://192.168.0.10).
If you have no DHCP Server available you can press F6 at boot screen and add additionally add the command parameter
static=xxx.xxx.xxx.xxx
netmask=xxx.xxx.xxx.xxx
bcast=xxx.xxx.xxx.xxx
gateway=xxx.xxx.xxx.xxx
dns1=xxx.xxx.xxx.xxx
dns2=xxx.xxx.xxx.xxx
where xxx.xxx.xxx.xxx is the IP of your choice.
By that VoiceBuntu boots with static IP address on eth0 network device.
Enhancements:
- This live CD release is based on Ubuntu 7.04 and VoiceOne 0.5.0_2 due to a minor bug in VoiceOne version 0.5.0.
Download (417.1MB)
Added: 2007-07-14 License: GPL (GNU General Public License) Price:
832 downloads
PlayVoIP 0.1
PlayVoIP is a webbased VoIP service enabler featured with user management. more>>
PlayVoIP is a webbased VoIP service enabler featured with user management, statistical report and registration application system based on LAMP and Asterisk.
At the beginning this project was specially designed for simple and free Indonesian VoIP network, VoIP Rakyat.
<<lessAt the beginning this project was specially designed for simple and free Indonesian VoIP network, VoIP Rakyat.
Download (0.027MB)
Added: 2005-12-02 License: GPL (GNU General Public License) Price:
1422 downloads
Linux LiveCD VoIP Server 2.0.23
Linux LiveCD VoIP Server can be used to provide a Vonage type service. more>>
Linux LiveCD VoIP Server can be used to provide a Vonage type service, or to create a voip pbx for a campus or business with up to thousands of SIP phones.
It is based on the Open Standard SIP Express Router (SER) and Asterisk. It can serve as a SIP Proxy, VoIP PBX, VoIP gateway or Class 5 Softswitch.
Main features:
- Easy Web user administration and real-time accounting.
- All in one solution to VoIP and SIP enable your business.
- Allows you to make your own SIP numbering plan. Centrex service.
- Can be connected to multiple A-Z wholesale termination providers and to your own PSTN termination gateway/router.
- Can do Least Cost Routing
- Includes nat traversal, stun server, media server for conference call bridge, voicemail to email, incomming virtual numbers (DIDs), follow me forwarding.
- Commercial pre-paid, post-paid and flat rate account support. No calling card (no b2bua in base system).
- Requires no software installation - it is a liveCD.
- Supports any SIP soft or hardware phones, such as popular XTen, Cisco ATA 186, Grandstream, Sipura, Bugetone, Linksys PAP2 and more.
- Supports SIP for Video Conferencing (Xten / CounterPath EyeBeam)
- Supports Ecrypted SIP with Xten / Counterpath Secure Xten Pro
- Requires a PC with fixed ip connection to the internet. 256 MBytes of RAM, CDRom reader, ide or sata hard disk to store call and user database and web site.
- Remote ssh configuration and administration help
Enhancements:
- Kernel 2.4.34.1 and minor bugfixes.
<<lessIt is based on the Open Standard SIP Express Router (SER) and Asterisk. It can serve as a SIP Proxy, VoIP PBX, VoIP gateway or Class 5 Softswitch.
Main features:
- Easy Web user administration and real-time accounting.
- All in one solution to VoIP and SIP enable your business.
- Allows you to make your own SIP numbering plan. Centrex service.
- Can be connected to multiple A-Z wholesale termination providers and to your own PSTN termination gateway/router.
- Can do Least Cost Routing
- Includes nat traversal, stun server, media server for conference call bridge, voicemail to email, incomming virtual numbers (DIDs), follow me forwarding.
- Commercial pre-paid, post-paid and flat rate account support. No calling card (no b2bua in base system).
- Requires no software installation - it is a liveCD.
- Supports any SIP soft or hardware phones, such as popular XTen, Cisco ATA 186, Grandstream, Sipura, Bugetone, Linksys PAP2 and more.
- Supports SIP for Video Conferencing (Xten / CounterPath EyeBeam)
- Supports Ecrypted SIP with Xten / Counterpath Secure Xten Pro
- Requires a PC with fixed ip connection to the internet. 256 MBytes of RAM, CDRom reader, ide or sata hard disk to store call and user database and web site.
- Remote ssh configuration and administration help
Enhancements:
- Kernel 2.4.34.1 and minor bugfixes.
Download (MB)
Added: 2007-03-25 License: Free To Use But Restricted Price: $399
636 downloads
VLOS 1.3.1 RC1
VidaLinux Desktop OS is a powerfull, stable and easy to use Linux distribution. more>>
VLOS is a powerfull, stable and easy to use Linux distribution. The main goal of Vidalinux Desktop OS is to make Linux Operating System easier to use and administrer using the right tools for your daily work on Home or Office.
Main features:
Poweful Desktop Linux!
- Beautful look and feel
- NFS network installation supported
- Hundreds of new software titles
- Faster boot-up time
- Enhanced Portage GUI
- Two CDs or one DVD installation
- PPC platform support
- gDesklets desktop applets
Hardware Support
- New 2.6.12 Kernel
- New KDE 3.4.1
- New Gnome 2.10
- New X.org 6.8.2 for improved video support
- New GCC 3.4.3
- New anaconda graphical installer supporting lastest graphics cards
- Support for SATA and PATA hard drives
Internet Applications
- Azureus Bittorrent Client
- GnomeMeeting Video conference and VOIP/IP
- Skype Instant messenger and VOIP/IP
- Gaim compatible with IRC, AOL, ICQ, Yahoo and MSN Messenger
- Xchat IRC client
- Firestarter enhanced firewall for added security
- Gftp FTP client
- Firefox Internet browser
Improved support for the Home entertainment
- New Amarok Music Manager
- New K3b CD, DVD, VideoCD burner
- New RealPlayer 10
- New Nexiuz First person shooter game
- New Trigger OpenGL rally car racing game
- Hydrogen music drum machine
Home and Office
- New Improved Wireless support
- Improved Networking, File Sharing and Printing
- New Nvu Web Authoring and editing tool
- New Evolution Groupware Suite (Reminders, MailMinder, Calendar, Address Book)
- New Openoffice 2.0 beta3
- New Beagle desktop searching tool
- Gnucash personal finance manager
Whats New in 1.3 Stable Release:
- Updated to gcc-4.1.1
- Updated to glibc-2.4
- Updated XGL and compiz-quinstorm
- Updated xorg-x11
- Fixed KDE problems with compiz
- Updated to latest baselayout
- Kernel updated to latest vidalinux-sources
- Update to portage-2.1.1-r1
<<lessMain features:
Poweful Desktop Linux!
- Beautful look and feel
- NFS network installation supported
- Hundreds of new software titles
- Faster boot-up time
- Enhanced Portage GUI
- Two CDs or one DVD installation
- PPC platform support
- gDesklets desktop applets
Hardware Support
- New 2.6.12 Kernel
- New KDE 3.4.1
- New Gnome 2.10
- New X.org 6.8.2 for improved video support
- New GCC 3.4.3
- New anaconda graphical installer supporting lastest graphics cards
- Support for SATA and PATA hard drives
Internet Applications
- Azureus Bittorrent Client
- GnomeMeeting Video conference and VOIP/IP
- Skype Instant messenger and VOIP/IP
- Gaim compatible with IRC, AOL, ICQ, Yahoo and MSN Messenger
- Xchat IRC client
- Firestarter enhanced firewall for added security
- Gftp FTP client
- Firefox Internet browser
Improved support for the Home entertainment
- New Amarok Music Manager
- New K3b CD, DVD, VideoCD burner
- New RealPlayer 10
- New Nexiuz First person shooter game
- New Trigger OpenGL rally car racing game
- Hydrogen music drum machine
Home and Office
- New Improved Wireless support
- Improved Networking, File Sharing and Printing
- New Nvu Web Authoring and editing tool
- New Evolution Groupware Suite (Reminders, MailMinder, Calendar, Address Book)
- New Openoffice 2.0 beta3
- New Beagle desktop searching tool
- Gnucash personal finance manager
Whats New in 1.3 Stable Release:
- Updated to gcc-4.1.1
- Updated to glibc-2.4
- Updated XGL and compiz-quinstorm
- Updated xorg-x11
- Fixed KDE problems with compiz
- Updated to latest baselayout
- Kernel updated to latest vidalinux-sources
- Update to portage-2.1.1-r1
Download (1418MB)
Added: 2006-12-02 License: GPL (GNU General Public License) Price:
1066 downloads
JVOIPLIB 1.4.0
JVOIPLIB is an object-oriented Voice over IP (VoIP) library written in C++. more>>
JVOIPLIB is an object-oriented Voice over IP (VoIP) library written in C++.
It is based upon work done for my thesis at the School for Knowledge Technology (or School voor Kennistechnologie in Dutch), a cooperation between the Hasselt University and the Maastricht University.
A part of this library was developed at the Expertise Centre for Digital Media (EDM) in Diepenbeek, Belgium. The EDM is a research institute of the Hasselt University.
Main features:
- Easy VoIP session creation and destruction.
- Highly configurable sessions: sampling rate, sample interval, compression type, ... can all be selected by the user. These features can also be changed during a session.
- Openness and extensibility: the object-oriented nature of the library makes it very easy to add features; new components can easily be tested by registering them as User Defined modules.
- Support for 3D effects: for my thesis I also did some research and development about VoIP in networked virtual environments, which included adding 3D effects to sound. For this reason, Ive added this feature to the library.
<<lessIt is based upon work done for my thesis at the School for Knowledge Technology (or School voor Kennistechnologie in Dutch), a cooperation between the Hasselt University and the Maastricht University.
A part of this library was developed at the Expertise Centre for Digital Media (EDM) in Diepenbeek, Belgium. The EDM is a research institute of the Hasselt University.
Main features:
- Easy VoIP session creation and destruction.
- Highly configurable sessions: sampling rate, sample interval, compression type, ... can all be selected by the user. These features can also be changed during a session.
- Openness and extensibility: the object-oriented nature of the library makes it very easy to add features; new components can easily be tested by registering them as User Defined modules.
- Support for 3D effects: for my thesis I also did some research and development about VoIP in networked virtual environments, which included adding 3D effects to sound. For this reason, Ive added this feature to the library.
Download (0.56MB)
Added: 2005-10-18 License: LGPL (GNU Lesser General Public License) Price:
1466 downloads
Yet Another Telephony Engine 1.2.0
YATE is a telephony engine aimed at creating an telephony server. more>>
Voice, video, data and instant messaging can all be unified under Yates flexible routing engine, maximizing communications efficiency and minimizing infrastructure costs for businesses.
Main features:
- VoIP server
- VoIP client
- VoIP to PSTN gateway
- PC2Phone and Phone2PC gateway
- H.323 gatekeeper
- H.323 multiple endpoint server
- SIP session border controller
- SIP router
- SIP registration server
- IAX server and client
- IP Telephony server and client
- Call center server
- IVR engine
- Prepaid and postpaid cards system
The software is written in C++ and it supports scripting in various programming languages (such as those supported by the curently implemented embedded PHP and Python interpreters).
Both PHP and Python libraries have been developed and made available in order to ease development of external functionalities for Yate.
Yate is production-ready software.
Yates license is GPL with an exception for linking with OpenH323 and PWlib (licensed under MPL).
Enhancements:
- This release adds support for Jingle (GoogleTalk), XML, and improved PBX support.
- It fixes one minor security bug and several others not security related.
<<lessMain features:
- VoIP server
- VoIP client
- VoIP to PSTN gateway
- PC2Phone and Phone2PC gateway
- H.323 gatekeeper
- H.323 multiple endpoint server
- SIP session border controller
- SIP router
- SIP registration server
- IAX server and client
- IP Telephony server and client
- Call center server
- IVR engine
- Prepaid and postpaid cards system
The software is written in C++ and it supports scripting in various programming languages (such as those supported by the curently implemented embedded PHP and Python interpreters).
Both PHP and Python libraries have been developed and made available in order to ease development of external functionalities for Yate.
Yate is production-ready software.
Yates license is GPL with an exception for linking with OpenH323 and PWlib (licensed under MPL).
Enhancements:
- This release adds support for Jingle (GoogleTalk), XML, and improved PBX support.
- It fixes one minor security bug and several others not security related.
Download (0.83MB)
Added: 2007-04-16 License: GPL (GNU General Public License) Price:
922 downloads
Gizmo Project Linux 0.8.0.13
An internet telephone as simple as an instant messenger. Freeware for linux more>> Gizmo Project is a free program that uses advanced VoIP technology to allow you to talk with your friends around the world.
Gizmo Project uses your internet connection (broadband or dial-up) to make calls to other computers. With the click of a mouse, you??e connected to friends, family, and colleagues anywhere on earth. It?? just that simple. You talk clearly. For as long as you want. For FREE.
Features:
Free Voicemail
Call Recording
Sound Blasts (sounds played while on the phone)
Instant Messaging
Call Mapping
Call Forwarding
Worldwide Conference Calls
SIP Network Peering<<less
Download (6.35MB)
Added: 2009-04-03 License: Freeware Price: Free
209 downloads
Voyage Linux 0.1
Voyage Linux is a Debian sarge-based distro (voyage) built from scratch. more>>
Voyage Linux is a Debian sarge-based distro (voyage) built from scratch. It is best run on a x86-based embedded platform such as Soekris 45xx/48xx and WRAP boards.
Main features:
- based on Debian Sarge r3.1
- 2.6.8.1 kernel
- prism54, hostap, madwifi, ipw2100, rt2400 drivers
- hostapd, wpa_supplicant from sarge
ToDo:
- improving installation scripts to allow different flavour for building customized distro
- scripts for setting up network configuration
- more wireless drivers (ipw2200, rt2500, etc.)
- further reducing in size
- light-weighted web server (thttpd + php) for system configuration
- bootable CD with voyage installer, pxeboot support
- more software features, like zebra/quagga, OpenVPN, FreeSWAN, traffic shaping/QoS, Asterisk/VoIP, etc.
<<lessMain features:
- based on Debian Sarge r3.1
- 2.6.8.1 kernel
- prism54, hostap, madwifi, ipw2100, rt2400 drivers
- hostapd, wpa_supplicant from sarge
ToDo:
- improving installation scripts to allow different flavour for building customized distro
- scripts for setting up network configuration
- more wireless drivers (ipw2200, rt2500, etc.)
- further reducing in size
- light-weighted web server (thttpd + php) for system configuration
- bootable CD with voyage installer, pxeboot support
- more software features, like zebra/quagga, OpenVPN, FreeSWAN, traffic shaping/QoS, Asterisk/VoIP, etc.
Download (52.5MB)
Added: 2005-10-17 License: GPL (GNU General Public License) Price:
1472 downloads
SILC Client 1.1.2
SILC is Secure Internet Live Conferencing. more>>
SILC (Secure Internet Live Conferencing) is a protocol which provides secure conferencing services in the Internet. It can be used to send any kind of messages, in addition to normal text messages.
This includes multimedia messages like images, video, and audio stream.
All messages in the SILC network are encrypted and authenticated, and messages can also be digitally signed. SILC protocol supports AES, SHA-1, PKCS#1, PKCS#3, X.509, OpenPGP, and is being developed in the IETF.
The software is delivered as SILC Client for end users, SILC Server for system administrators, and SILC Toolkit for application developers.
Enhancements:
- This version fixes several crash bugs and other minor bugs.
<<lessThis includes multimedia messages like images, video, and audio stream.
All messages in the SILC network are encrypted and authenticated, and messages can also be digitally signed. SILC protocol supports AES, SHA-1, PKCS#1, PKCS#3, X.509, OpenPGP, and is being developed in the IETF.
The software is delivered as SILC Client for end users, SILC Server for system administrators, and SILC Toolkit for application developers.
Enhancements:
- This version fixes several crash bugs and other minor bugs.
Download (2.1MB)
Added: 2007-07-05 License: GPL (GNU General Public License) Price:
842 downloads
OSP Toolkit 3.4.0
OSP Toolkit project is a client side implementation of the ETSI OSP VoIP Peering protocol (ETSI TS 101 321). more>>
OSP Toolkit project is a client side implementation of the ETSI OSP VoIP Peering protocol (ETSI TS 101 321).
The OSP Toolkit project was begun in 1998 and the code has been incorporated into many commercial and open source VoIP products.
<<lessThe OSP Toolkit project was begun in 1998 and the code has been incorporated into many commercial and open source VoIP products.
Download (0.41MB)
Added: 2007-05-08 License: BSD License Price:
901 downloads
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