How Are You Facilitating Your Inbound and Outbound DID’s?

April 3, 2008 by Garrett Smith

Direct Inward Dialing and how it works

Many of us may have questions about how DIDs work and how to provision them. DID stands for “Direct Inward Dialing”. DIDs are typically used in conjunction with an IP PBX, to route incoming and outgoing calls to their correct source or destination. Almost every IP PBX has a method of facilitating DID’s, whether that be internal or external to the server. However, products used to facilitate them are often in question.

There are three ways that most businesses are “bringing in” their DID’s. The first method is Analog Trunking or “POTS” (Plain Old Telephone Service). Analog Trunks may be comprised of physical copper PSTN lines paid for and supplied by your local telephone company. These are pure RJ-11 analog lines, no different from your “landline” wall jack at home. Most SOHO applications utilize analog POT’s lines since they are more cost effective. The typical number of physical PSTN lines is usually around four to eight, and will vary depending upon the number of inbound/outbound calls the business needs to support. Each physical POTS line is equal to one channel, and represents a 1:1 ratio. Each physical PSTN line also has a single DID number associated with it. If you have four analog POTS lines, you have four DID’s or channels available to make inbound and outbound calls.

Removing Voip from the picture for a second.

Take VOIP out of the picture for a second…. in a true analog environment, each PSTN line would be connected to an analog telephone (user), and that telephone would be associated with a specific DID number. Let’s bring back VOIP now….analog telephone lines are NOT connected to the phones themselves, since in most cases, you will be using VOIP phones, but rather into the central location… the IP phone system. Analog lines are facilitated within the IP PBX via FXO PCI cards. If you have eight incoming RJ-11 PSTN lines, you will essentially need an eight-port RJ-11 analog PCI card much like the Sangoma A20004D or Digium AEX808E. Simply connect each RJ-11 connection into the ports on these cards, most IP PBX’s will auto-detect their presence, and you are now permitted to configure your analog trunks or channels within the IP PBX. Since the analog DID’s are now facilitated at a central level, they are not on a pure 1:1 basis because when a VOIP phone accesses this trunk group to make an outbound call, it is not associated to that one DID always, it will simply grab the next available DID within the channel group and use that. This allows users to add more VOIP phones to the scenario without physically increasing their number of POTS lines. Essentially, if you have 16 VOIP users, but only experience around 8 concurrent calls at a time, you would only need 8 POTS lines, rather than 16. Please note, you are not limited to four or eight-port analog PCI cards. These numbers were offered as a very basic example. Please check out Sangoma and Digium on voipsupply.com for further clarification on these cards.

Digital connections are becoming very popular amongst larger organizations because of ease of use and cost savings over Analog POTS Trunking. In most large applications, there is a need for 24, 48, or even 96 + voice channels. The easiest method to facilitate this number of voice channels is via digital T1 lines using a T1 provider. Most IP PBXs have the ability to integrate digital T1 connections, either through T1 digital PCI cards or external T1 gateways. A T1 is an essential 24 individual lines (equivalent to 24 analog POTS lines) delivered over a single pipe. A T1 is configured at the IP PBX level through digital trunk groups or channel groups.

2 Types of T1 Provisioning

T1’s can be provisioned in two flavors. The first is through PRI signaling (Primary Rate Interface). PRI’s contain 24 channels in total but only allow for 23 configurable channels. The remaining channel is a “work-horse” channel so to speak, performing all of the overhead signaling work for each of the 23 available channels. The second method of T1 signaling is a method called CAS (Channel Associated Signaling). CAS signaling with the “T” allows for 24 channels to be configured as voice, data, or both. Each channel performs its own work to allow for proper signaling to take place. Please check with your T1 provider to ensure which method they are using, and opt for the best method to fit your needs. Digital T1 lines are integrated with an IP PBX very easily….simply connect the T1 to a Sangoma A101 card or Digium TE-122P card, which are both Single T1 cards. However, you are not limited to a single T1; a standard digital PCI card can have up to four T1 ports incorporating 96 channels within the system, but that doesn’t mean you can’t add a second quad T1 card. For those larger T1 applications I just spoke about, please refer to the Sangoma A104D card or the Digium TE-420B.

SIP Trunking

The final method to bring in your DID connections is through a method called SIP Trunking. SIP Trunking is done completely over a data connection (Typically T1, Fiber, ADSL or Cable), and is provisioned to your PBX through a connection to the WAN and your SIP Trunk provider. SIP Trunking is becoming one of the most cost-effective methods of acquiring inbound and outbound channels because there is no need for a physical connection on the premises. The only down-side? Certain providers will only provide local SIP Trunks to specific geographic locations. Check with your prospective VoIP provider for availability of SIP Trunking services in your area. SIP Trunking is quickly gaining in popularity with businesses of all sizes. I think you will see more providers start to offer this solution, if they don’t already, and we should see SIP Trunking to almost every local DID as time and demand progress. Everything nowadays seems to be “up in the clouds.” SIP Trunking is no different, and is conforming to the future specifications of what consumers, both large and small, are expecting.

New Redfone Communications Product Alert

April 2, 2008 by Garrett Smith

Redfone Communications New Releases

We have added two new Redfone Communications products to our ever expanding Voip Supply Catalog. The first is the foneBridge2-EC Single which is a T1/E1 PRI-to-Ethernet Bridge. It is an integrated black box “appliance” designed to streamline installation and enable redundant design of Asterisk based VoIP/PBX systems. The second is also a RedFone Communications offering. It is a foneBridge2 Single T1/E1 Bridge Asterisk PBX T1/E1 Redundancy, High Availability & Load Balancing in an economical Enterprise class solution. As always you can either purchase online or call one of our friendly sales engineers with any further questions you may have.

VoIP Supply Unleashes Go3 Guaranteed Replacement Warranty Program

by Garrett Smith

Go3 allows customers an affordable way to protect their hardware investment

Go3 Guaranteed Replacement Warranty

VoIP Supply announced the launch of its new three-year, no-questions-asked, guaranteed replacement warranty, Go3. The new Go3 Guaranteed Replacement Warranty gives customers a no hassle way to protect their VoIP hardware purchases. The new warranty program is based off of VoIP Supply’s existing warranty program and was redesigned with feedback from existing customers.

“Our customer service is VoIP Supply’s number one priority,” said Benjamin P. Sayers, CEO of VoIP Supply. “We have been successful due to our wonderful customers’ trust and support. The Go3 Guaranteed Replacement Warranty is just another way that we can bring our clients needs to the forefront.”

The Go3 Guaranteed Replacement Warranty allows customers to protect their VoIP Supply hardware investments. Their hardware is completely covered for a full three years from the date of purchase. If the equipment fails, malfunctions, or ceases to operate properly for any reason (manufacturer defect, component failure or user negligence), VoIP Supply will provide a replacement.

Unlike most extended warranty programs offered by retailers, with Go3 there is no fine print or run-around during the redemption process. The only conditions include purchasing the equipment from VoIP Supply and returning it within the three-year period. The coverage must also be purchased at the point of sale, and it is not refundable.

For more information about the VoIP Supply Go3 Guaranteed Replacement Warranty, please visit the go3 warranty for more information.

What does Value Added mean exactly?

March 31, 2008 by Garrett Smith

Quality customer service at VoIP Supply

As a consumer, how many times have you been in a situation wondering the status of a good or service being provided to you? It’s not a great feeling to be left in the dark, so why do consumers feel the need to live in the “dark”? The excuses of inadequate time, lack of patience, or simply the unknown meaning of great representation, seems to be present daily throughout the minds of many consumers. Because of this, I want to reach out to you, let you know we are here to help, find solutions, answer questions, keep you happy, follow through, but most of all–make a friend!

Providing customer service at all levels is something VoIP Supply prides itself in, and it’s improving daily. What aspect of a business is more rewarding then having a satisfied, excited customer? How do we actually do this? Providing a live body, excellent support, pre/post sale support, and many additional add-ons has not only become the focus of VoIP Supply, but is at the core of our work ethic. As a growing company, the focus to fill your needs is our top priority, and it shows on a daily basis.

What greater reward could you ask for? Let our staff turn the “lights of service” on for you, and help you become the “excited” consumer. As your care taker we want your hospitality, your presence, and dedication. My final line to you is to simply try us; I guarantee you will be in for a pleasant surprise. Let us be the “Value-Added” to your day.

Asterisk Appliance Roundup

March 28, 2008 by Garrett Smith

Asterisk-based, IP PBX software platforms has lead to demand for inexpensive, telco-grade hardware appliances

The proliferation of open source, Asterisk-based, VoIP PBX software platforms has lead to demand for inexpensive, telco-grade hardware
appliances. There are likely more than a dozen manufacturers who offer dedicated server / appliance hardware, in a variety of form factors, for use with popular platforms including Asterisk, Trixbox, Elastix and others.

Many Asterisk users and integrators choose to run their PBX on off the shelf Intel or AMD-based servers from the likes of Dell and Supermicro, while neither of these manufacturers target the Asterisk marketplace per se. Here’s a quick overview of some of the hardware choices out there if you are looking to deploy open source telephony in your home or business.

Arizona-based Rhino Equipment caters specifically to Trixbox users.

In addition to manufacturing TDM hardware interface cards for Asterisk, Rhino offers several appliances. The Ceros Mini is their latest offering, released a few weeks ago. The Ceros Mini is Intel based and occupies a 1U form factor, with a 2GB flash based hard drive, 512MB RAM and two usable PCI slots for PCI cards. The Ceros Mini also has an integrated LCD display and five-button keypad. Rhino offers a variety of upgrade options from the base configuration.

The Rhino Ceros is the big brother to the Mini, occupying a 3U form factor and sharing many of the same features as the Mini. The Ceros offers a wider range of HDD and RAID options, as well as more usable PCI and PCI Express slots for higher density applications. Both Rhino appliances ship standard with Trixbox loaded.

Relative newcomer Rockbochs

Hailing from Duluth, Minn., Rockbochs is a relative newcomer in the marketplace, offering their Phonebochs Telephony Appliance which ships with the latest stable version of Trixbox business VoIP PBX software. Like the Ceros Mini, the Phonebochs occupies a 1U form factor, but with a more industrial configuration of CPU, RAM and HDD. An Intel Core Duo Mobile CPU, 1GB RAM and RAIDed 80GB SATA hard drives, and three onboard 10/100/1000 NICs are standard, plus a number of upgrade options.

Fonality released its Trixbox Appliance

California’s Fonality released its Trixbox Appliance last year. The Trixbox Appliance comes in a standard and enterprise version. Intel P4 CPU, 512MB RAM and 80GB hard drive, and integrated four-line LCD display round out the package in a 3U form factor.

Pika Technologies

Telephony industry veteran Pika Technologies has thrown its hat in the ring with the release of its diminutive Warp Appliance for Asterisk. Featuring a compact form factor and AMCC Power PC 440EP Embedded Processor, the Warp Appliance packs a ton of features into a small footprint, and can be configured with a variety of TDM hardware FXS/FXO combinations for PSTN and legacy analog hardware integration.

Aastralink Appliance

Aastra Telecom has also thrown its hat in the ring; having announced its Aastralink Appliance at the recent Spring Von show.

Digium released its Asterisk Appliance

Finally, Georgia’s Digium released its Asterisk Appliance in 2007, and has already announced an OEM partnership agreement with telecom mainstay 3Com. The Digium Asterisk Appliance offers a compact, solid state form factor with hardware-based echo cancellation and up to eight analog FXS/FXO ports onboard. Digium also recently announced the Switchvox AA60, a small footprint PBX appliance designed to host its popular Switchvox SOHO and SMB Software.

Traditional telephony vendors such as Cisco Systems, Nortel and Avaya continue to struggle in making their products accessible, manageable and affordable to the small to midsized business crowd, leaving the door wide open for hardware and software vendors such as those mentioned above.

WTFM – Write the Fantastic Manual

March 26, 2008 by Ben Sayers

Previously I have written about the value of reading the manual early on in your exploration and use of a new application, toy, tool or otherwise. However boring the material is, the advanced reading will almost always save you significant time trying to make things work and will reduce your overall level of frustration.

In any business, someone needs to write the manual before others can read it. If you are fortunate to be in a position of being first, you are also likely to be challenged with this task. This is so often overlooked by small companies, particularly young ones in rapidly changing environments. This step is also frowned upon by those with any ounce of creativity in their job as they often feel that they are artists and their role cannot possibly be documented.

1. Why write the manual, I already know how to do my job?

2. I’m too busy to write the manual, how about I get to it when things slow down?

3. I’m an artist; you don’t expect me to document how I create!

4. Why should I take the time to write the manual when there are so many more important things to do?

The only valid question is the last one, and there are so many great answers to it. The first answers are easy:

1. Do you already know how to do your job? Prove it. Besides, if you want a promotion you need to train and backfill your position, wouldn’t that be easier and faster with process documentation?

2. If you are successful, things should not “slow down.” They should just get more efficient. Document how to become more efficient and thereby, more successful.

3. If you get hit by a bus tomorrow, can I easily find your past creations and pick up on open tasks where you left off?

4. There are dozens of great reasons for documenting what you do throughout the day and how you complete each task. Below are some of the reasons why.

a. Documenting each task helps to plan your day today and tomorrow. Having a plan, even if it is little more than a task list, will help keep you focused and ensure that you are able to complete all of your objectives each day/week/month. Document to plan, plan to succeed.

b. Proper documentation will allow you to take vacation with less stress. Knowing that someone can pick up a book and walk through the steps to complete your job in your absence should be a load off your mind. Document to relax.

c. Every process has room for improvement and areas to focus on becoming more efficient. By documenting each process and the steps to complete it, you have drawn it out and are able to evaluate it from the outside in, something not possible when you are already in the middle of the task. Document to improve.

d. By having a complete listing of your tasks and the individual steps to completion is a huge highlighter on your professionalism. Taking the time to understand what you do, to share that knowledge and maintain it as business changes, makes you an invaluable asset to the company. Not doing so make you a liability. Document to build value and professionalism.

e. Keeping internal job secrets might seem like a good way to protect your job and ensure employment security. The opposite, at least at my company, is the actual truth. By not documenting you become a liability and need to be replaced with someone who is secure through professionalism and results not by keeping secrets. Document to build job security (and keep your job).

f. If you are in management, how do you hold your staff accountable? Did you train each one individually? Do you shadow them daily? By documenting their processes, you can build in measurements and points to ensure proper execution. Your staff’s results fall on your shoulders. Knowing what they do and how they do it allow you to measure and to hold your staff accountable. Document for knowledge and success.

Like many companies, we started small and fast with little documentation and a constant stream of changes. We still move quickly and change a lot, but we also realize the value of stopping to look around, documenting what we do, and evaluating it for purposes of improvement and overall business knowledge. However boring, tedious and unnecessary the documentation process may seem, the value of the output is second to none in relation to effective and efficient business operations.

The Future of Internet Protocol

March 21, 2008 by Garrett Smith

Hello again! This is your friendly Tech Guy, Kyle Brocious, here.

Internet Protocol and its evolution

Today I would like to talk about a subject that many of us probably have not started to think about. That subject is on Internet Protocol and its evolution. As many IT people will know the current Internet Protocol known as IPv4 has been around for almost 25 years. You know the ins and the outs of how it works and its limitations, however, for those of you that are not of the IT world (don’t be ashamed, there are many like you) in a nut shell, it is the basis to which the internet works. Every address you type into your web browser has an IPv4 address. Let me break this down just a little more. When someone types in www.voipsupply.com they are in fact typing in 192.168.50.33. Go ahead try it out. You can copy and paste the numbers or type it in, and it will lead you right to our site. Now isn’t that cool?

By 2010 the face of the internet will have to change

Now what only a handful of people have started to contemplate is how much longer these IPv4 address will be around. Just think about it. There are only so many combination’s of numbers that can be used when putting in that format. Some Estimates say that by 2012, 17 billion devices will be online, that is accounting cell phones, laptops, computers, PDA’s, music players and a slew of other gadgets. Looking at it from that angle, there is only one third of IPv4 addresses left.

Now before you may start to worry about the internet crashing, back in 1990 the IT world banned together and started to think of a solution on how to fix this rising shortage of internet addresses. After tossing around some ideas the concept of IPv6 was born. This idea was the basis of many studies and is the future of the internet.

When migrating to IPv6 the address size jumps dramatically from 32-bit with IPv4 to 128-bit with IPv6, which would allow about 18 quintillion people their own set of 18 quintillion addresses (3.4e38 total addresses). With that many addresses I do not see any issues with running out for a long, long time.

But the drawback of the large address size is that IPv6 carries some bandwidth overhead over IPv4, which may hurt regions where bandwidth is limited. This is where header compression can sometimes be used to alleviate this problem. And IPv6 addresses are also very difficult to remember, but it is possible to use of the Domain Name System (DNS) if necessary.

However, migration has proved to be a challenge in itself. The reason being is not all of the networking equipment in the world is IPv6 compatible. Now it may be possible for some of the VoIP equipment to get a firmware update to which will make it work, but that will only work for about 15 to 25 percent of network equipment around the world.

So what I ask of you is to start to think about what will be needed for this migration, the implementation, and monetary cost of the expansion. It is not a matter of If at this point, but a matter of when. My personal estimate will be that by 2010 the face of the internet will have to change, and with this change will be many companies and individuals purchasing IPv6 compliant equipment. Supply and demand law says that the higher the demand the higher the cost, so it may be a wise choice to start looking into what you will need to make this possible.

Treating Griddle-Forehead Syndrome (GFS) with VoIP Technology

March 20, 2008 by Garrett Smith

Isn’t it wonderful e-mailing friends all over whenever you want, wherever they are? Who hasn’t e-mailed a cousin in Europe, a friend in Pismo Beach, or Piscataway, or Piston Springs?

I know I have.

Now how much did that cost you for all the e-mails and instant messages you could stuff into a 30-day period? Plus, your family members or roommates use these services too.

I would say you probably spent anywhere from $10 to $50 depending on how fast you like to “talk” with dial-up or high speed internet.

REWIND 20 YEARS…In June 1988, my parents opened the local Telco bill on check-writing Sunday and saw the normal $37.00 bill had jumped to $406.00. 406 Big Ones! My father, usually a laconic man, used several choice words to berate me that day when he saw the phone bill…with 50 or so calls going to my girlfriend in Florida. You could have fried an egg on his forehead.

I know I’ve already posed several questions to you, but are you a business owner, world traveler, telecom director, or angry parent having griddle-head moments like Phil Sr. did 20 years ago, today?

FAST-FORWARD to 2008…
Twenty years and 37 pounds later…
• Boston Red Sox win two World Series
• Arnold is Governor of the biggest state in the Union
• Al Gore invents the internet (have another one…)

Seven years after that fateful Sunday two things happened–I started e-mailing my girlfriend, and my father began recognizing me again.

Today, think of that $406 going towards a new VoIP deployment. All of the phone exchange with whoever you want or need to talk to is at internet pricing–whether it is a relative, customer, vendor or girlfriend. Your days of the huge Telco bill are over; should you decide to accept the mission of saving money and avoiding griddle-forehead syndrome, or GFS.

Simply compare your phone bill to your ISP bill. Which is more? How much do you pay to talk to a friend outside the county you live in? How about state or country? Have you noticed those rates have decreased, but still your bill is taxed, re-taxed, state taxed, federally-taxed, surcharged and so on?

Multiply this by the number of employees you have using phones at your business. Sorry to remind you about that touchy subject, but that is because the Telcos are advertising lower rates to reel you in and then charging you wherever else they can to get the $406+ smackers. Why else would they lower their rates? Isn’t that the American way?

Take a half-hour out of your day and call one of the experts here at VoIP Supply. Your success is paramount to ours. FOLLOW US!

New VoIP Hardware At VON; Coming to VoIP Supply’s Shelves Soon

March 18, 2008 by Garrett Smith

Benjamin Sayers, Cory Andrews, Arthur Miller and I are all here at the Spring VON show in San Jose this week and today we got a chance to tour the exhibit hall floor and check out some of the new hardware that will be coming to the shelves at VoIP Supply in the next few weeks.

Here is a rundown of the products that should quickly become top sellers:

1. Linksys WIP310 – This a new SIP based WiFi phone from Linksys. It comes with a slick color screen and it 802.11/g compliant.

2. SwitchVox AA60 Appliance – Digium owned SwitchVox announced their new SwitchVox appliance which will replace their tower design that is currently in distribution later this month. It is designed to meet the needs of the small business and seems to be their response to the trixbox appliance.

3. Aastra Astralink 160 – Based on a Asterisk, Aastra has released their own phone system that seems to replace their own VentureIP phone system that was very popular a few years back. Basic in features, much of the call control and configuration is done on the handset, not the phone system. Set-up to deployment in about 5 minutes.

4. IP EVO Xing – Old Skype hardware, new strategy. According to the folks a IP EVO the once Skype only USB conferencing solution will be re-released in the next two months as an pen SIP based product. When it was available, it was a popular item. Look for demand to quadruple when it comes in an “open” version.

5. Zyxel SIP DECT – Zyxel had two different SIP DECT solutions on hand at their both. Neither of them where yet available, but they looked slick and had robust feature sets. More to come on this.

Look for more from us on these exciting new products in the very near future.

Take Advantage of VoIP Technology

by Garrett Smith

A close look at a simple Point to Point VoIP adapter solution from Linksys that could save you millions on those dreaded long distance fees.

We have all encountered them; hidden fees. Those spikes in the phone bill are the reasons why we have to question, “Where did that charge come from?” or “I didn’t realize that was a long distance call!” Want to put a stop to all of that? You would be foolish if you didn’t, so take a look at this solution:

VOIP Technology is driven through a high speed internet connection. Calls are mostly made (in this scenario) over the SIP protocol, unlike the traditional POTS (Plain Old Telephone System) where calls pass over a series of circuited switched networks owned by your Telco provider. So let’s drop the Telco provider out of the scenario and focusing on setting up a free point to point setup.

First you are going to need the following:

• A working high speed internet connection at both locations
• A Linksys PAP2T at each location
• An analog telephone to connect to the PAP2T’s in each location.
• This instruction guide for proper setup…

Linksys spa-ata and PAP2T Point to Point Instructions

The following diagram below will lay out the setup for this process and all instructions and screenshots will reference it.
Linksys spa ata schematics diagrams

SITE A Configuration

Instructions
1. Connect an analog telephone to phone 1 port on a VoIP adapter unit.
2. Connect an Ethernet cable into the LAN port on your unit and power on.
3. From the analog phone, press ****, then 110# to get the IP Address of the unit.
4. Open your internet search browser from a computer and browse to that IP Address in the address bar.
5. Upon entering the Linksys Web GUI Configuration, click the System Tab.
6. Make the following changes.

Here is a typical point to point scenario:

Let’s say I will be traveling to Europe in the upcoming months, but I would still like to remain in contact with my friend in Florida. I am fully aware that if I call her from a European land line, I will most likely incur long distance fees, which will inevitably add up over time. But before I went to Europe, I did some shopping on VoIPSupply.com and purchased two Linksys PAP2T’s with the full intention to setup a pure VOIP solution to make free calls to and from my friend in Florida over the World Wide Web.

After receiving the PAP2T’s, I properly configured each unit to the specifications noted in the instructions above. I chose whatever extension worked best for me, and ensured that each would have a static IP address. This is a very important aspect of this setup since the two units “talk” to each other via IP and are “registered” to one another to form this IP communication link. After I had configured both units to “point” to one another, I made test calls to each, and the calls were made to and from both units on a pure VOIP basis.

This solution costs around $100 and takes about 30 minutes max to setup. The best part about the solution is you can bring the PAP2T anywhere, connect it to a high speed internet connection and make free calls to your other point. This is a point to point solution so you are only able to call the other recipient your device is configured to, and are limited to those boundaries.