It combines the flexibility of advanced call routing found in today's VoIP PABX systems with the security of multiple failover systems to bring considerable cost savings and peace of mind

Ranging from the Pure Edition with pure VoIP, to the Pro with Analogue and VoIP right through to the Enterprise Edition supporting up to 350 channels. The ConduIT Pro and Pure editions are based on AstLinux and Asterisk™ and were unveiled by the Prime Minister of New.Zealand in May 2007.

Some Key Features:

New Zealand ConduIT Customers

The ConduIT VoIP PABX is a revolution in telecommunications.

Supports both VOIP and Analogue Video Conferencing Audio Conferencing Extensions 1 to Thousands Fax support All Call Forward options Call Waiting Caller ID for Call Waiting Off Premises/IP Stations Admin Web Management Interface Last Call Redial Do Not Disturb Voice Mail Forward to e Mail Hold Call Park & Park Pickup Call

All Major ADC functions IVR Support Advanced Call Reporting Advanced Time Rule Management Wake-up Call Traffic Shaping VPN Support (optional) Multi-Tenant Toll Barring Dial Direct from Outlook, Thunderbird Night Bell Codec Transcoding All major Hunt Groups options Screen Popping Call forwarding

• Screen Popping
  
  Screen popping from a number of Open Source applications mean that you can know who is calling you before you answer.
  
  

• Click to Dial
  
  Click to dial applications allow you so simply click on a person's details in order to call them.
  
  

• Combine multiple offices into one answering point.
  
  If you have staff directing calls from multiple current locations across a company, you can combine all of this into one place but still allow for extra calls to go to other numbers.
  

• Lower Call Costs
  
  Because the ConduIT Systems use the VentureVoIP Exchanges for selected calls, you are able to gain the significantly lower call costs when using out Least Cost Routing (LCR).
  
  

• Free VentureVoIP Calls
  
  Any calls to someone who is connected to the VentureVoIP system are free. This includes extensions in remote offices and even on other companies' systems.
  
  

• Free audio and video conferencing
  
  Video and audio conferencing is free on the VentureVoIP exchanges when used with approved video conferencing equipment.
  
  

• GSM/3G Cellphones
  
  You can include your cell phones as extensions of your system using GSM or 3G Cellular trunking units.
  

Because of the way the ConduIT system is designed, there are failsafes at every point.

• Multiple Exchanges
  
  VentureVoIP has multiple exchanges located both around New Zealand and internationally at peering exchanges.
  
  All ConduIT PABX systems are constantly connected to all exchanges at the same time. If any one of the connections between your ConduIT and our exchanges fails, it will remove that connection until it is working properly again. So, if for example Auckland was struck by a natural disaster, your calls would be instantly routed via Christchurch.
  
  This also means that you don't end up with the PABX vendor blaming the provider, as we are the provider!
  
  

• Analog failover
  
  If your Internet connection fails you and you have no connection to any of our exchanges, you can still make and receive calls through normal telephone lines connected to the system.
  
  

• Configuration Changes
  
  Any time that you change a setting on your ConduIT, it is stored in the VentureVoIP exchanges. This means that if you have an absolute distaster at one location (i.e. fire, earthquake, flood), you can simply get a new device, add the username and password to it, and it will download all of your settings to the new device.
  
  This allows your company to be back up and running in as small a time as possible.
  

The ConduIT is a collection of software and hardware developments by VentureVoIP and others which combines the best of technology with a simple user friendly interface.

Core components include software from VentureVoIP's ConduIT3 controls, Asterisk™, AstLinux, lcdproc, busybox and others.

• Traffic Shaping
  
  The ConduIT IPBX sits between your Internet connection and the rest of your network. By prioritising packets and slowing downloads you can good quality stable VoIP calling. This is a requirement on residential connections, and adds quality to business environments.
  
  We were previously using the AstShape script which was based on the wondershaper script, but have recently switched to Gurney Halleck's improvement of Maciej Blizinski's HFSC script.
  
  

• Packet Loss Concealment
  
  This is a feautre which originally was only supported by certain codecs, but is now also available for standard codecs in Asterisk™.
  
  The WikiPedia has this to say:
  
  Packet Loss Concealment (PLC) is a technique to mask the effects of packet loss in VoIP communications. Because the voice signal is sent as packets on a VoIP network, they may travel different routes to get to destination. At the receiver a packet might arrive very late, corrupted or simply might not arrive. One of the situations in which the latter could happen is where a packet is rejected by a server which has a full buffer and cannot accept any more data. In a VoIP connection, error control techniques such as ARQ are not feasible and the receiver should be able to cope with packet loss.
  
  

• Bandwidth Requirements
  
  It depends on what compression codec you use, but we normally use either GSM (compressed) or ALAW (uncompressed).
  
  The amount of bandwidth required also drops per simultaneous call because we use IAX trunking.
  
  For example using GSM:
  
  1 Call Incoming bandwidth: 27.84 Kbps
  1 Call Outgoing bandwidth: 27.84 Kbps
  
  2 Calls Incoming bandwidth: 40.84 Kbps
  2 Calls Outgoing bandwidth: 40.84 Kbps
  
  10 Calls Incoming bandwidth: 144.84 Kbps
  10 Calls Outgoing bandwidth: 144.84 Kbps
  
  And using ALAW:
  
  1 Call Incoming bandwidth: 78.84 Kbps
  1 Call Outgoing bandwidth: 78.84 Kbps
  
  2 Calls Incoming bandwidth: 142.84 Kbps
  2 Calls Outgoing bandwidth: 142.84 Kbps
  
  10 Calls Incoming bandwidth: 654.84 Kbps
  10 Calls Outgoing bandwidth: 654.84 Kbps
  
  Those figures are in Kilobits per second (I.E. a 2Mb connection is 2048Kbps).
  
  So, assuming you only have 1 concurrent call and it is in GSM, you would use 27.84Kbps x 60seconds = 1670.4Kb. This is Kilobits, not KiloBytes.
  
  To get the figure in KiloBytes, you divide by 8:
  
  1670.4Kb / 8 = 208.8KB
  
  So, again assuming only one simultaneous call, using GSM, you are looking at around 5 minutes of call per megabyte. So if you had a 2GB cap on your connection, you would be able to have 9,578 minutes of calls or 159 hours. This would increase if some of the calls are simultaneous.
  
  

• Network Address Translation
  
  Because we are using the High Quality Inter Asterisk™ Exchange protocol (IAX2) for calls, you don't have to forward media ports like you do with SIP and OpenH.323. This even works with the IAX softphone or our VentureVoIP IAX Phones. A registration is made from your system to ours regularly and hold open a single port which is used for both calls and control. Because of IAX2 trunking, all the calls are multiplexed into a single connection which reduces bandwidth overheads because of less packet overhead.
  
  Please note that external people using SIP phones instead of an IAX phone may need to do port forwarding.
  
  

• Asterisk™
  
  Asterisk™ is an Open Source communications platform started early in 1999, just after the Linux Expo. At the time, Digium was still "Linux Support Services, LLC" which was Mark and a group of contractors. He had done a development job for Adtran (who shared the booth with me at the show) in which we used a Linux box with a frame relay card to act as a frame relay to ethernet bridge for this DSL mux called a "frameport". Keith Morgan, who was representing Adtran at the event had brought some Atlas boxes to show off some of their telephony stuff and he thought "Hey, what happens if Keith sends me a voice over frame relay call?" so he tried that and he got a little blip of data. He theorized that if he could get a call into the PC he could do anything with it, and thus was Asterisk™ born. He needed a phone system anyway and with as small a startup budget as he had for LSS, he wasn't about to buy one, so building one seemed a logical way to go.
  
  Mark decided to make available all of the source code for Asterisk™ under the GPL Open Source licence. This meant that straight away, anybody could contribute to Asterisk™, building it into the product it is today.
  
  The VentureVoIP staff have been working closely with Asterisk™ for a number of years and have their own patches included as part of the distribution that you can download.
  
  For more information on Asterisk™, you can visit the about page on the Asterisk™ web site.
  
  

• Linux
  
  The WikiPedia has this to say:
  
  Linux (IPA pronunciation: /'l?n?ks/) is a Unix-like computer operating system family. Linux is one of the most prominent examples of free software and of open source development; its underlying source code is available for anyone to use, modify, and redistribute freely.
  
  After the Linux kernel was released to the public on 17 September 1991, the first Linux systems were completed by combining the kernel with system utilities and libraries from the GNU project, which led to the coining of the term GNU/Linux. From the late 1990s onward Linux gained the support of corporations such as IBM, Sun Microsystems, Hewlett-Packard, and Novell.
  
  Predominantly known for its use in servers, Linux is used as an operating system for a wider variety of computer hardware than any other operating system, including desktop computers, supercomputers, mainframes, and embedded devices such as cellphones. Linux is packaged for different uses in Linux distributions, which contain the kernel along with a variety of other software packages tailored to requirements.