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ST-PBX Live 0.7.5

ST-PBX Live 0.7.5


ST-PBX Live is a GNU/Debian Linux distribution integrating Asterisk. more>>
ST-PBX Live is a GNU/Debian Linux distribution integrating Asterisk.

FreePBX and other tools to try from CD or install in HD a complete VoIP system.
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Download (306.5MB)
Added: 2007-01-02 License: GPL (GNU General Public License) Price:
1031 downloads
SimplePBX 0.0.3

SimplePBX 0.0.3


SimplePBX is a configuration/dial plan manager for Asterisk. more>>
SimplePBX is a configuration/dial plan manager for Asterisk. Supports multiple trunks and call groups. This is a Multi-Tenant system requiring very little configuration to have a fully working PBX system. Complete GPL source code.

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Download (0.051MB)
Added: 2007-03-19 License: GPL (GNU General Public License) Price:
949 downloads
PBXware SOHO 1.8.0

PBXware SOHO 1.8.0


PBXware is a scalable PBX solution. more>> PBXware is a scalable PBX solution featuring a range of traditional telephony and emerging VOIP technologies. The creation of national/global voice networks and a full range of PSTN and VOIP technologies is supported. Least cost routing, Enhanced Voicemail, ACD Queues, IVR Auto Attendants, Conference Bridges , Music on Hold and much more... a real cost saving in a fully featured PBX solution!
PBXware is delivered in CD, tarball, appliance or VPS formats each supporting an easy to use Setup Wizard allowing setup of a fully functional PBX in minutes. Administration is performed through included Web Interface or CLI. Auto updates, system backup, provider templates, call recordings, real time call monitoring are just some of advanced features included.
PBXware offers flexibility, performance, delivery methods, technologies, easy to use yet extremely advanced features not found at any competitors offerings. Should you found same features set elsewhere we would like to hear from you.
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Download (700.00MB)
Added: 2009-04-09 License: Freeware Price: Free
205 downloads
Solid PBX SVN-trunk-r152

Solid PBX SVN-trunk-r152


Solid PBX is a multi-platform Soft Switch Class 5 software targeted for home, corporate, and operator level markets. more>>
Solid PBX is a multi-platform Soft Switch Class 5 software targeted for home, corporate, and operator level markets.
GETTING STARTED
First, be sure youve got supported hardware (but note that you dont need ANY special hardware, not even a soundcard) to install and run Solid PBX.
Supported telephony hardware includes:
- All Wildcard (tm) products from Digium (www.digium.com)
- QuickNet Internet PhoneJack and LineJack (http://www.quicknet.net)
- any full duplex sound card supported by ALSA or OSS
- any ISDN card supported by mISDN on Linux (BRI)
- The Xorcom AstriBank channel bank
- VoiceTronix OpenLine products
The are several drivers for ISDN BRI cards available from third party sources. Check the voip-info.org wiki for more information on chan_capi and zaphfc.
Enhancements:
- This is a major bugfix release with a simplified chan_h323 build process and many improvements.
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Download (9.9MB)
Added: 2006-05-17 License: GPL (GNU General Public License) Price:
1255 downloads
AdminsParadise VoIP PBX 1.0 Beta

AdminsParadise VoIP PBX 1.0 Beta


AdminsParadise VoIP PBX is a full Web-based phone and fax solution more>>
AdminsParadise VoIP PBX is a full Web-based phone and fax solution that integrates the best-of-breed open source VoIP software.
It runs Asterisk 1.4.1, hylafax, avantfax, and PHP5 with a themed, easy-to-use, Web-based interface.
Main features:
- Asterisk 1.4.1
- HylaFax
- AvantFax
- Scheduled Conferencing
- Integrated authentication
Enhancements:
- This initial release provides a live CD and an installation CD.
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Added: 2007-03-20 License: GPL (GNU General Public License) Price:
951 downloads
Solid PBX 20060616 (Sounds-en)

Solid PBX 20060616 (Sounds-en)


Solid PBX is a multi-platform Soft Switch Class 5 software targeted for home, corporate, and operator level markets. more>>
Solid PBX project is a multi-platform Soft Switch Class 5 software targeted for home, corporate, and operator level markets.
GETTING STARTED
First, be sure youve got supported hardware (but note that you dont need ANY special hardware, not even a soundcard) to install and run Solid PBX.
Supported telephony hardware includes:
- All Wildcard (tm) products from Digium (www.digium.com)
- QuickNet Internet PhoneJack and LineJack (http://www.quicknet.net)
- any full duplex sound card supported by ALSA or OSS
- any ISDN card supported by mISDN on Linux (BRI)
- The Xorcom AstriBank channel bank
- VoiceTronix OpenLine products
The are several drivers for ISDN BRI cards available from third party sources. Check the voip-info.org wiki for more information on chan_capi and zaphfc.
Enhancements:
- Sound files and music on hold files supplied for use with Solid PBX have been replaced with new versions, and are available in two languages (English and French) and in five formats (WAV (uncompressed), mu-Law, a-Law, GSM, and G.729).
- In addition, the music on hold files provided by FreePlay Music are now available in the same five formats, but no longer available in MP3 format.
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Added: 2006-06-17 License: GPL (GNU General Public License) Price:
1225 downloads
CosmoPBX 1.0 / 2.0 Alpha

CosmoPBX 1.0 / 2.0 Alpha


CosmoPBX is Linux live CD based on Knoppix Linux. more>>
CosmoPBX project is Linux live CD based on Knoppix Linux. CosmoPBX is released as open source under the GNU General Public License (GPL) and is available for download free of charge. With CosmoPBX, it is possible to get a PBX system up and running within 2 minutes without installing anything! CosmoPBX is designed to reduce the time spent in installation and setup of typical PBX system with Asterisk.
CosmoPBX is built from Knoppix which has solid Debian Linux foundation. CosmoPBX is bundled with latest Asterisk 1.4.0 and Asterisk-GUI Beta. With Asterisk and Asterisk-GUI installed, one doesnt have to worry about installing Operating System, dependencies and all necessary packages to get the IP PBX system running with Asterisk PBX.
Whats New in 1.0 Stable Release:
- This is the first stable version of installable CosmoPBX Live CD. This version has several performance enhancements as well as some bug fixes. This version of CosmoPBX Live CD can be installed to local hard disk with simple installation wizard. Feel free to report bugs and feature requests to sourceforge forums and trackers.
Whats New in 2.0 Alpha Development Release:
- Java SSH, SugarCRM, Flash Operator Panel, PHPMyAdmin, Webmin for Server Control, A2Billing, support for saving configuration data to hard drive or other media when running from a live CD, a PHPInfo page, a backup and restore feature, a Web GUI for Asterisk CLI, the ability to view syslog, the ability to view asterisk log files, the ability to shutdown and reboot the server, the ability to control asterisk from a GUI, support for changing date/time, and support for changing timezone were added.
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Download (605MB)
Added: 2007-06-28 License: GPL (GNU General Public License) Price:
849 downloads
AdminsParadise Voip PBX and FAX 1.0

AdminsParadise Voip PBX and FAX 1.0


AdminsParadise Voip PBX and FAX is an enterprise Class VoIP PBX and Fax server that features Asterisk and Hylafax and more... more>>
AdminsParadise Voip PBX and FAX is an enterprise Class VoIP PBX and Fax server that features Asterisk and Hylafax and more...

VoIP Can offer a significant savings for a small, medium or large office. Free enterprise grade VoIP PBX and web based fax solution. Features extensive "movie walkthroughs" to step you through the installation and administration of the software.

LiveCD and installation CD Features: Easy web based administration. Rock solid platform. Industrial Grade Hylafax Fax solution with web based faxing and print-to-fax capability Asterisk VoIP engine. Easily schedule conference bridges with an intuitive web interface. Modular design. As the administrator, you have the ability to choose which modules your users have access to. Roaming users.

Your users can login to any phone and their number follows them. Easy web based configuration, and more! Again the purpose of Adminsparadise is to provide administrators with the best-of-breed Open Source solutions in a manner that is extremely easy to use and administer. Enterprise Feature set to include call parking, paging, Interactive Voice Response, music-on-hold, custom queuing and much much more Use any open standards based phone (SIP) to include Polycom, Cisco, Grandstream, Snom, Aastra, and more. Easy to install, administer and free

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Download (701.9MB)
Added: 2007-04-07 License: GPL (GNU General Public License) Price:
563 downloads
sipX 3.6.0

sipX 3.6.0


sipX is a next generation IP PBX solution offering rich functionality combined with ease of use. more>>
sipX is a next generation IP PBX solution offering rich functionality combined with ease of use, installation, and administration. sipX is entirely based on the Session Initiation Protocol (SIP) and significant attention is paid to standards compliance and interoperability.
It combines all common calling features, XML-based SIP call routing, Web-based configuration, and integrated management and configuration of the PBX and attached phones and gateways.
It is a modular server-based solution that does not require any additional hardware, as it interoperates with any SIP compliant gateway, phone, or application.
Main features:
- Native SIP IP PBX Solution
- Voicemail
- Multiple auto-attendants
- Presence
- SIP call routing
- Web based management
- Scalable architecture
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Download (22MB)
Added: 2007-07-20 License: LGPL (GNU Lesser General Public License) Price:
849 downloads
SIPp 2.0

SIPp 2.0


SIPp is a free Open Source test tool / traffic generator for the SIP protocol. more>>
SIPp is a free Open Source test tool / traffic generator for the SIP protocol. It includes a few basic SipStone user agent scenarios (UAC and UAS) and establishes and releases multiple calls with the INVITE and BYE methods.
SIPp project can also reads custom XML scenario files describing from very simple to complex call flows. It features the dynamic display of statistics about running tests (call rate, round trip delay, and message statistics), periodic CSV statistics dumps, TCP and UDP over multiple sockets or multiplexed with retransmission management and dynamically adjustable call rates.
Other advanced features include support of IPv6, TLS, SIP authentication, conditional scenarios, UDP retransmissions, error robustness (call timeout, protocol defense), call specific variable, Posix regular expression to extract and re-inject any protocol fields, custom actions (log, system command exec, call stop) on message receive, field injection from external CSV file to emulate live users.
While optimized for traffic, stress and performance testing, SIPp can be used to run one single call and exit, providing a passed/failed verdict.
Last, but not least, SIPp has a comprehensive documentation available both in HTML and PDF format.
SIPp can be used to test many real SIP equipements like SIP proxies, B2BUAs, SIP media servers, SIP/x gateways, SIP PBX, ... It is also very useful to emulate thousands of user agents calling your SIP system.
Enhancements:
- New: Statistical (conditional) branching feature. See
- http://sipp.sf.net/doc/reference.html#Randomness+in+conditional+branching.
- New: 3PCC extended mode - see
- http://sipp.sourceforge.net/doc/reference.html#3PCC+Extended
- Tool: monitor remote SIP servers through SNMP - see
- http://sipp.sourceforge.net/wiki/index.php/Getting_feedback_from_the_server
- Enh: extends the -aa option to UPDATE messages
- Enh: changes in the compilation with external libs - useful for the package
- generation system
- Enh: Allow sampling from a Weibull distribution for pause duration
- Enh: use stat_delimiter for the trace_rtt option and include number that is
- being reported.
- Enh: Add repeat_rtd keyword for repeated RTD calculations
- Enh: Handle stripping Control-M characters from multi-valued headers in
- get_header
- Enh: Update the clock_tick more frequently so that we have a higher timer and
- statistics resolution
- Enh: for option that take a time as argument - allow them to be specified using
- seconds or milliseconds
- Enh: fail when parsing a scenario that has pcap if pcap is not enabled
- Enh: print the actual location of the error log file and the error condition
- (if any) on creation
- Enh: fail when parsing a scenario that enables authentication if SSL is not
- enabled
- Enh: Makefile - include EXTRAENDLIBS keyword so that libraries can be appended
- to the list after SSL Enh: Add regexp_match argument to the receive command for
- universal catching
- Enh: remote control: increase the allowed number of control sockets.
- Enh: 3pcc extended: clean the reach of the allowed number of local twin sockets
- Enh: amelioration of statistic computing
- Enh regexp: add case_indep, occurence and start_line options for the hdr
- matching case
- Enh: Stats: Use RTDs that are precise to the microsecond in -trace_rtt, and
- improve the consistency between trace_rtt and the averages
- Enh: Only include RTDs that are actually used in the CSV output
- Enh: Allow loss percentages less than 1 and also a global command line option to
- specify that packets should be lost at a given percentage
- Fix: fix for -t un error for non-IPv6 platform like win32 - Unable to bind UDP
- socket, errno = 106 (Address family not supported by protocol)
- Fix: Do not initialize the screen library in background mode
- Fix: allow having an optional recv before a recvCmd
- Fix: Authentication: allow [fieldn] values for the [authentication] field
- Fix: updated support of short header forms
- Fix: small bug fix to the micrortt.diff which is required for the initial call
- rate to work properly
- Fix:Allow the code to compile with -Wall -Werror on Linux
- Fix: empty line was generated when [routes] keyword was used and proxy did not
- record-route
- Fix: pcap on HPUX; Fix: Simple fixes identified with valgrind
- Fix: 3pcc extended: problem when quitting
- Fix: regexp: add a warning when the specified header is not present in the
- received message
- Fix: pcap: destroy the send packets thread properly even if the sendto failed
- Fix: bug in regexp due to an incomplete commit (rev 172) - remove some debugging
- traces in message log file
- Fix: bugs on retransmission counters and on cookies for optional messages
- Fix: Incorrect branch in automatic ACK answering to unexpected >= 400 responses,
- as well as automatic CANCEL - in such cases, branch must be identical to the
- branch of the initial INVITE request
- Fix: 3pcc extended: clean up when screen exit. Fix: stop logging when the
- maximum allowed file size is reached (avoid core dump)
- Fix: incomplete Via header in automatic responses when aborting calls
- Fix: -h: -key parameter
- Fix: 3pcc/3pcc extended modes: closes twin sockets properly when twin instance
- exits, to break the poll() loop and avoid the print of empty messages
- Fix: in 3pcc/3pcc extended modes: send BYE/CANCEL before exit due to other twin
- instance exit
- Fix: force exit when pressing q twice (Q press can still be used)
- Fix: Aka authentication for synchro case, also added password len as password
- for authentication might contain char Fix: possible core dump in SDP parser
- Fix: accept up to the 5 defined RTDs (previously would only accept one less)
- Fix: Fail if there is an invalid repartition table specification - previous
- behavior was a core dump
- Fix: added -users option to the parameter table
- Fix: change option description to match timed options
- Fix: trace_err did not work in background mode
- Fix: bug when testing the presence of the 3PCC compilation flag
- Fix: bug in -r -rp option
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Download (0.18MB)
Added: 2007-04-27 License: GPL (GNU General Public License) Price:
926 downloads
ZoIP 0.2.1

ZoIP 0.2.1


ZoIP is a bridge between telephony and text adventure for Asterisk and Infocom games. more>>
ZoIP project is a bridge between telephony and text adventure for Asterisk and Infocom games.

ZoIP is a Rezrov Z-Machine for the Asterisk PBX, allowing you to play Infocoms old interactive fictions, such as Zork and The Hitchhikers Guide to the Galaxy, over the telephone using speech-to-text (sphinx2) and text-to-speech (festival).

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Download (6.5MB)
Added: 2006-12-19 License: Perl Artistic License Price:
1045 downloads
SIP Express Router 0.9.6

SIP Express Router 0.9.6


SIP Express Router is a very fast and flexible SIP (RFC3261) server. more>>
SIP Express Router (ser) is a high-performance, configurable, free SIP ( RFC3261 ) server .
SER features an application-server interface, presence support, SMS gateway, SIMPLE2Jabber gateway, RADIUS/syslog accounting and authorization, server status monitoring, FCP security, etc. Web-based user provisioning, serweb, available.
Its performance allows it to deal with operational burdens, such as broken network components, attacks, power-up reboots and rapidly growing user population.
SERs configuration ability meets needs of a whole range of scenarios including small-office use, enterprise PBX replacements and carrier services.
Main features:
- accounting
- digest authentication
- CPL scripts
- ENUM support
- instant messaging
- MySQL support
- PostgreSQL support
- a presence agent
- Radius authentication and accounting
- Diameter authentication
- record routing
- SMS gateway
- Jabber gateway
- NAT traversal support transaction module
- registrar
- user location
SER has been extensively and successfuly tested with many SIP products from other vendors (Microsoft, Cisco, Mitel, snom, Pingtel, Siemens, and many others). It has been powering our SIP services continuously for more than two years.
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Download (2.0MB)
Added: 2006-01-11 License: GPL (GNU General Public License) Price:
1393 downloads
Kiax 0.8.51

Kiax 0.8.51


Kiax is an IAX client application (a so called Softphone) which allows PC users to make ordinary VoIP calls to Asterisk servers. more>>
Kiax is an IAX client application (a so called Softphone) which allows PC users to make ordinary VoIP calls to Asterisk servers, the same way as they do it with their hardware telephone.

Kiax aims to provide a simple and user-friendly graphical interface and desktop integration for calling, contact list, call register management and easy configuration. That is - a simple to use IAX Client.

VoIP has become a hot topic and already is a part of our life. Different standards and platforms are already in development and use: H.323 and SIP are only some of the most famous used VoIP protocols. A marvellous piece of software called Asterisk PBX (http://www.asterisk.org), developed by Mark Spenser from Digium Inc. (http://www.digium.com) proved again that the Open Source CAN PRODUCE QUALITATIVE SOFTWARE for the business and can save money with its full set of features of an advanced PBX.

In the process of the development of Asterisk a new protocol has gathered the attention among the VoIP users - the Inter-Asterisk eXchange, or IAX (TM), Protocol, used as a the native communication protocol between Asterisk PBX Servers. What is particularly good in it is that it requires only one UDP port per endpoint to create a successful communication channel for VoIP calls. This makes it much friendlier for users behind NAT, which is not the case with SIP and H.323 - they required specific router configurations for their channel establishment, thus hindering faster spread of VoIP on the home and corporate desktop.

The IAX protocol is already adopted by major VoIP providers and Telcos. If you register in Free World Dialup (http://www.fwdnet.net), IAXTel (http://www.iaxtel.com) or Freshtel (http://www.freshtel.net) or any other VoIP termination service like VoIPUser.org (http://www.voipuser.org) or PSTN Termination Provider like VoIP Jet (http://www.voipjet.com) then you may be interested in using Kiax.

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Download (0.40MB)
Added: 2006-06-08 License: GPL (GNU General Public License) Price:
1244 downloads
StrongBox Linux 1.0 beta14

StrongBox Linux 1.0 beta14


StrongBox is a next generation Linux-based operating system. more>>
StrongBox is a next generation Linux-based operating system, incorporating tight version control, digital signatures, high security and built-in change management.
StrongBox represents a shift in the Linux paradigm. Moving away from monolithic systems where everything runs together, StrongBox is a fully modular Linux distribution.
It incorporates a small ram-based OS and toolset for administration and security functions, and then allows applications bundles, built using any Linux distribution, to run on top of it. Each of these bundles, running in a secure virtual context, has a large level of independence from the base OS.
This secure partitioning allows StrongBox to have some unique and powerful features. OS versioning, rollbacks, automated hardware detection, fault-tolerant boot process, easy migration from server-to-server, and the ability to incorporate other Linux distributions and existing installations are key features.
StrongBox gets its name from the high standard of security that it provides. Using virtual servers, digital signatures on all OS components, and a large number of read-only pieces in the OS, StrongBox is highly resistant to security breaches.
Also, it has virtual boot medium independence, meaning that it can run with little or no changes from CDRom, compact flash, USB, SCSI, etc. This makes StrongBox ideal for the secure deployment of embedded systems, computing clusters, enterprise Linux deployments, secure web servers, PBX systems, and kiosk systems.
Enhancements:
- makerpm and makedebian scripts greatly improved
- asteriskathome build script
- Improved usb authentication, and automatic key selection for USB keys
- Added a gcc, distcc, linux headers, and kernel source add-ons
- Unionfs, bind, vrenice, rpm, hashlimits subsystem
- Larger default config partition (12MB)
- dmraid support re-added (use dmraid kernel parameter on boot)
- Updated versions of all software
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Download (465MB)
Added: 2005-09-01 License: GPL (GNU General Public License) Price:
1513 downloads
KObelisk 1.01

KObelisk 1.01


KObelisk is a tool that connects to the Asterisk PBX and gives information to the user about incoming calls. more>>
KObelisk is a tool that connects to the Asterisk PBX and gives information to the user about incoming calls. KObelisk also provides some basic functions like calllists, ringtones or a callback function.

Installation:
1) use ./install.sh (as root) to install after this you will find a KObelisk entry in your menu

2) start the program

3) configure KObelisk by rightclicking on the little KObelisk Icon at the traybar (most important is the Asterisk module configuration)
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Download (0.16MB)
Added: 2006-06-26 License: GPL (GNU General Public License) Price:
1215 downloads
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