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Mark Spencer on Skype and Asterisk

Aaron Rosenthal — Mon, 02 Aug 2010 16:15:42 +0000

Here’s an interesting video from earlier this year with Mark Spencer addressing Skype and Asterisk.

America 2009 Video: Skype and Asterisk – What it Means for Business Communications (Mark Spencer)

Writing an RFP for an Open Source IP PBX – Part2

Aaron Rosenthal — Fri, 18 Sep 2009 15:40:31 +0000

Welcome to my second installment of “Writing an RFP for an Open Source IP PBX”… But first let me apologize to my loyal readers, I have let you down… I’ve been caught up in the regular grind of the summer and have neglected OS-VoIP. There’s a lot of interesting things going on in the Open Source VoIP space and I’ve got a long list of things I want to talk about, so get ready for some action here at OS-VoIP! Ok… onto RFP’s…

In my previous post, here, we went over a variety of reasons why Open Source needs to be approached and perceived differently than regular proprietary systems. We also discussed the importance of determining (in detail people!) your own requirements as the very first step in this process.

Part deuce starts with this………

Answer the important questions

Regardless of where you place this information in an RFP, it should always be addressed. The earlier you make these decisions the easier it will be to start off on the right foot when asking vendors to quote you an Open Source IP PBX solution. This is by no way a complete list of questions but these are the important ones.

Do we plan to have a converged or segregated voice and data network?
This is a very important decision and will sometimes dictate the type of phones a vendor recommends as part of your solution. For example, if you are converging the voice and data networks and in turn daisy chaining desktop computers to the phones, you’ll want a phone with a switch, and possibly GigE support should GigE to the desktop be important. Some phones don’t even have a switch so be careful not to make the mistake of converging your networks only to find that the 150 cheap Polycom 320’s you purchased don’t have switches in them! Most network engineers would recommend that the reliable approach to take is keep your voice and data networks separate but this sometimes carries a greater upfront expense such as running a second ethernet drop to each workstation or purchasing twice as many switches.

On the topic of wiring, I’m going to share a suggestion which some of you might disagree with. Should you ever need new wiring, DO NOT make it a requirement in your RFP that your OS PBX vendor also be required to handle wiring! My opinion – office wiring should be left up to electricians… I’ve always said that any company (specifically in the OS IP PBX world) who does both wiring and Open Source IP PBX’s is going to lack significant skills in one area or the other. Good engineers who really know how to program software like Asterisk sit at their computers all day being programmers… they don’t usually make the best wiring contractors. We live in a world of specialization, companies can either do twice as much half ass’ed, or half as much bad ass! Wiring and software development are just too different for most companies to want to maintain dedicated resources for both.

What type of phone service do we want to use with this new Open Source IP PBX?
The beauty of most OS IP PBX systems is that they will work with almost any type of phone service. A lot of people for some reason think that an IP PBX will only work with digital phone service and this is not the case. If you are replacing a legacy phone system then chances are you’re using either a T1/PRI or POTS lines which won’t have any problem plugging into your new OS IP PBX. Now a vendor isn’t going to care much whether you stick with a PRI or POTS because that usually won’t impact the overall solution (other than the type of telephony cards/gateways quoted), BUT if you plan on using other types of phone service, then there’s a few more things a vendor should address and therefore why you need to let the vendor know about these in the RFP.

So what am I referring to? SIP and Internet based VoIP service. Most large businesses wouldn’t consider using internet based VoIP for their primary voice service because of call quality issues that still plague these options (no QoS, latency, etc), so I won’t discuss these here. Dedicated/private SIP on the other hand is a very good option for an OS IP PBX system. Dedicated/Private SIP is where the phone company is actually providing you with a dedicated voice circuit, such as a T1/fiber/ethernet, which is delivering SIP service directly from their network to your office. Using your data connection to run SIP over the internet is NOT a private SIP solution, it is a public SIP product. Here’s a little article I wrote about SIP which I hope to elaborate on soon.

One of the main advantages of using a SIP service is that you offload call transcoding from the PBX resources onto the carriers network. Asterisk for example is SIP native, so is FreeSWITCH, so by using SIP as your primary phone service that call can in many cases ride un-altered from the handset all the way through the carriers network and the transcoded to the work on the receiving end. By eliminating the need for the PBX server to transcode the call, you save resources and in-turn increase the call capacity of your server.

How do you plan to administrate your Asterisk system?
This is probably the most important question to be answered because what you say to this question will entirely dictate the type of Asterisk flavor you purchase. Although building an Asterisk system using open source or BE Asterisk can result in the most reliable, fully functional, and flexible PBX solution (when engineered properly), it also lacks what most IP PBX buyers want to see which is a user and admin interface. Asterisk is entirely programmed and configured within the command line, and unless your company is of a size where you can afford to hire a telecom administrator with Asterisk experience, you will probably need some sort of administration interface. If you are of the size where having a dedicated employee/s to manage your PBX… then it makes perfect sense to hire Sys. Admins with Asterisk and Linux experience. Actually, if you have these resources then you have even more of a reason to build an OS IP PBX. Remember that if you require an admin interface, and are a company larger than a couple hundred users, then most of your Asterisk PBX options will be licensed products. If you want that admin interface instead of employing an Asterisk engineer then you’re going to likely get tied into licensing and per seat pricing models… just FYI.

Existing PBX

It’s very useful for a vendor to know what type of PBX you’re currently running. This should give your vendor at least a general overview of the features and functionality current users are comfortable with, plus the vendor should be familiar enough with other PBX systems to know what features may or may not work differently to those found in an OS IP PBX.

Within your RFP, list the features of your existing PBX which are currently USED by your employees. Everyone utilizes different features so be sure to include all your departments in the conversation while you uncover all those features you never even used but others might find quite handy. This is a good beginning to your requirements gathering process so don’t forget to also prioritize which existing features are important and which ones are not.

Call Centers

Now here’s a hot Open Source topic… Asterisk and Open Source software for the call center-
I’m not going to stray down this road too far because writing an RFP for a call center PBX is usually quite different than a standard enterprise communication system. Call Centers have about a gagillion feature requirements and in many cases these requirements are so specific that a highly customized Asterisk system is sometimes just what the doctor ordered. Typically in a call center, specifically one using Open Source technology, Asterisk is rarely the only software being used as part of the “total solution”. There’s a whole slew of applications and OS software which… when combined… can rival even some of the most established call center solutions.

Actually, not many people knew this but Genesys (dubbed the worlds #1 contact center software) used to fully support Asterisk. Quite a few call centers were deployed with Genesys software running in conjunction with the Asterisk PBX. I think this is a fairly good testament to Asterisk’s increasing presence in the large enterprise space. Genesys even used to be a participating partner in the Asterisk community… but not any more. Word on the street is Genesys partnered up with Microsoft with an integrated solution with Microsoft communication products and that was pretty much the end of their partnership with Digium… dang it.

Proprietary call center applications are purchased in modules – inbound, outbound, agent portal, etc….and each module costs more money. In the Open Source VoIP space, modules are replaced by 3rd party applications which work in conjunction with software like Asterisk. For example, ViciDial is free software for Asterisk designed for outbound call centers. QueueMetrics is licensed software with hundreds of reporting functions and options that works in conjunction with Asterisk. Aheeva makes a fully functional call center solution built entirely on Asterisk but unfortunately they’ve licensed the product out the wazoo which I think is very non-open-sourcian of them…

If you are compiling an RFP for an Open Source call center, then you definitely take your phone system more seriously than most businesses… obviously… because it IS your business. And naturally you’re concerned with the risk you might take with implementing an Open Source PBX system. As a call center I’m sure writing an RFP is as familiar as cotton candy at the carnival, so I won’t draw this out more than I need to. BUT… what I will say is “keep an open mind”. Your biggest risk in implementing Open Source VoIP software for a call center is not the technology itself, it’s the capabilities and experience of the company who implements the solution. Or, depending on your own technical chutzpah, maybe you can develop your whole call center PBX internally. I recently lost a call center bid for a 400+ agent facility due solely on the fact that the non-technical CEO hadn’t heard of Asterisk… and despite the fact we met EVERY requirement, and despite being able to scale infinitely with their growth, and even despite the capital cost being about $250,000 less than Cisco… they still got cold feet due to inaccurate assumptions and a bloated stomach from all the proprietary spoon feeding.

Here’s an important tip about the RFP process- if you’re making a recommendation to the CEO or CIO to purchase an Open Source IP PBX… insist that they participate in all discussions with the vendor. I’ll be the first to say that it’s not a simple task to understand these systems, and reading a proposal (the result of your RFP) is rarely going to single-handedly sell an OS IP PBX against a proprietary system. If a CIO is reading two proposals, one by Cisco for example, the other an Asterisk system… their instant gut reaction to the Asterisk proposal is to completely discredit it because the price is so low. If the decision maker isn’t educated about these systems, seeing a 50% price difference usually hurts more than it helps. They think “oh this solution must be the Yugo of phone systems” or “this vendor has wildly underestimated our requirements” or “where’s the other zero” ….

Without fully understanding the technology, the development, the capabilities of these systems… a CIO can not make an educated decision about an Open Source IP PBX. What is the solution, I’ve already said it… participate in discussions with the Open Source vendors, don’t make assumptions, ask questions when you have concerns, and LISTEN to the answers.

When buying Cisco or Avaya… the primary characteristics a decision maker cares about are features and price; they usually don’t question the products ability to simply “work”. Features and price are simple to address in a proposal. An Open Source IP PBX on the other hand… decision makers question EVERYTHING, will it be reliable? Will I get support? Does it have the features I need? Will it be too hard to use? These are all gut reactions which must be addressed by the vendor, and it’s no easy task. You simply can’t eliminate all these concerns through information in a single proposal… no matter how large it might be… It requires numerous calls, meetings, and research on the part of the final decision maker. Unless the final decision maker takes the initiative to understand these products, and participates in calls with the vendors to explain these systems, then they’re not going to be equipped for the questions and concerns they’ll undoubtedly have once a solution has finally be recommended by the vendor. So why should the CEO, CIO, or final decision maker spend so much time in understanding these systems…. because if an OS IP PBX is the right fit, they stand to save their own company A LOT of money, not just in upfront capital but most importantly in ongoing costs.

Solutions you might need, but didn’t know could be done with Open Source VoIP

And lastly I want to leave you all with a slightly greater understanding of the capabilities of software like Asterisk… because once you have a well engineered Open Source IP PBX, you’ll learn there’s much more you can get out of that bugger, the only limitation is… well…you. When it comes to the world of technology, and software, and especially voice, if you can think it… you can do it with Open Source software (caveat-if you know what you’re doing) For example, I’m working with a company right now for whom we built a custom IVR solution which in a lot of cases would be the extent of this systems capabilities. But with Asterisk, we are now able to take the same hardware, same software, and with some additional development increase the capabilities of their solution from an IVR to a full blown PBX to support almost one hundred remote call center agents.

So, where you might think you need multiple solutions or systems… you’ll find that sometimes your highly customizable Open Source IP PBX will scale and do the job just fine.

Here are some examples:

Paging – I see this ALL the time, I’m sure you do too. Company has their PBX, and they have their dedicated external paging system, and the two integrate. Well because endpoint costs on OS IP PBX’s are so low, I see very little reason why you wouldn’t use your PBX as your paging server as well. Paging features such as multi-zones, two way paging/intercom, and music are features which can all be replicated by most PBX systems. You can either use analog paging speakers, or IP paging speakers…. check out CyberData’s product. They have a very impressive product line, let me just forewarn you that every time I’ve ordered their products they’ve showed up anywhere from 2-6 weeks late. External paging with Asterisk = works swimmingly. All this is on top of the obvious fact that you can still page through the handset.

Emergency Notification – Companies can spend LOTS of money on emergency notification systems… but a lot of the same functionality can be dealt with by Asterisk. In an Emergency, the idea is to send out blasts to a particular group using all communication methods… email, voice, text, page, etc. By integrating Asterisk with something like a contact database, you could initiate Asterisk to make outbound blasts to phone numbers… either record a message or send via text which Asterisk can then read via a text to speech engine. On top of that, integrate Asterisk with an SMS gateway and that’ll take care of texting. Additionally, Asterisk could page all phones and paging speakers… and with the addition of a basic web app or custom script I’m sure there’s a LOT More you can do… like tweet, or facebook, oh the possibilities!

Access Control - Did you know that with IP access control devices which are SIP compliant, you can actually use Asterisk for access control? The interface might be a little dodgy but let’s say you have IP card readers to access a building, Asterisk can actually open the door (integrated with a door jam – analog or IP), you can log that action just the same as you can log when a user makes a phone call, you could even program Asterisk to make a call once someone entered a particular room. For example, maybe campus or corporate security receives a notification or call from Asterisk whenever someone enters a particular secured area. There are big massive giganticly expensive security systems that do just this, and usually people will integrate their PBX with said systems… but if your requirements are fairly basic, and you’re willing to compromise on sexy interfaces, you’ll find that your PBX might do exactly what you need for a fraction of the cost. Plus you eliminate the integration costs.

Now I know what you’re thinking… “if I use my Open Source IP PBX for everything won’t it become a single point of failure?” It all depends on how the system is built, maybe you build dedicated Asterisk systems with a single function. Or, with ALL the cash dollaz you save by using your Open Source PBX to replace other dedicated proprietary systems, you’ll have some extra budget to beef up the redundancy of your IP PBX. Pay a little extra to cluster or load balance your systems, purchase larger and more reliable servers, beef up your network redundancy, buy more POTS lines for fail over, heck… get some internet SIP trunks, and keep the PRI… You can make any technology completely fault tolerant with 100% up time… you just have to make the right development decisions and you need to spend the right amount of money. The lesson here is what would be cost prohibitive in the proprietary world is often affordable when evaluating an Open Source IP PBX.

Keep your options open, start from an “anything is possible with Open Source” mentality… and work backwards from there.

If you’re in the market for more than just a PBX, discuss with an Open Source PBX vendor and perhaps include a list of your requirements in the RFP. Just maybe you’ll kill multiple birds with one stone… a stone meticulously made out of Open Source software and COTS hardware!

Oh, and one last thing. Building an Open Source IP PBX is NOT free… I get this a lot. For some reason people expect to just buy the hardware, install the software and BAM, they’re good to go…. wrong. Large enterprises should realize that if you’re looking at a proprietary system that costs a million dollars for example, an Open Source system may cost half that. Nevertheless we’re talking real money, and even at half the price, five hundred thousand is still a chunk of change. If you plan on doing an Open Source IP PBX the right way, plan on paying top dollar for the best hardware and the best development resources. But you know what’s awesome….even if you pay top dollar for everything… your capital investment is still going to be a heck of a lot less than what you’d expect to pay for a proprietary system… and you’ll probably get a heck of a lot more!

Good luck hunting…

By Aaron Rosenthal

An Asterisk Milestone – Shove 10,000 simultaneous calls onto a single Open Source machine!

Aaron Rosenthal — Fri, 28 Aug 2009 20:54:29 +0000

Recently Olle E Johansson posted some details about how he managed to get 10,000 simultaneous calls out of a single Asterisk server. As far as I’m aware, this is the largest number of simultaneous calls documented on a single Asterisk based server, so congrats Olle… I hope you enjoy that bottle of wine!

Now who says you can’t get more than 250 calls on a single Asterisk server??? I’ve always known that with the right setup and configuration you could get at least 2,000 calls on a single Asterisk server… but 10,000 is a remarkable milestone. It’s news like this that further validates the significance Asterisk and even other open source VoIP software plays in the world of carrier communications. If I were a company like Cisco or Sonus, I’d pay very close attention to all this.

Now yes, 10,000 calls is a lot.. but it is inevitable and as Asterisk continues to evolve, its ability to handle more and more calls will increase. The maximum number of simultaneous calls which Digium (creators of Asterisk) will support is 250 calls, I really hope that soon they increase this capacity because it really is stifling Asterisk’s growth amongst service providers… but then again, these type of large call loads are not easy to achieve and often require the assistance of those who are extremely well versed in Asterisk.

From reading the post, I did get the feeling that there was a little bit of voodoo involved with reaching this benchmark, but hopefully we’ll see some documentation soon to follow so others can start replicating and testing such a load on large servers.

Premise based Open Source IP PBX or hosted VoIP?

Aaron Rosenthal — Tue, 08 Jul 2008 21:54:37 +0000

This is a question I hear constantly and the answer typically relies on three things – cost, business size, and required functionality. Most small-mid sized businesses exploring the move to a new phone system will consider the ups and downs between purchasing a premise based IP PBX or a hosted VoIP solution. These days I think purchasing an old TDM phone system instead of an IP based system is about as smart as buying a VHS player so lets assume that most businesses won’t go this route. Cost, reliability, scalability, and functionality each vary depending on which solution your business chooses.

In true OS-VoIP fashion, I’ll make the comparison between hosted VoIP and an Open Source IP PBX because there are some unique advantages an OS IP PBX has over hosted VoIP. I’ve seen plenty of companies decided that hosted VoIP made more sense than a proprietary premise based IP PBX but once you throw an OS IP PBX into the mix, the metrics can change quite dramatically.

Cost:

A customer premise OS IP PBX will require an upfront capital expense plus the additional cost of yearly maintenance which can vary in price depending on the level of maintenance required. Although an OS IP PBX does not hold the same costly licensing fees associated with proprietary systems, the capital expense is still not one to overlook. Depending on the type of OS IP PBX, you’ll probably discover a savings of 30%-50% compared to its proprietary counterpart.

By using a hosted VoIP service you’ll avoid this capital investment but may soon find that the operating expense usually exceeds the capital cost of a customer premise OS IP PBX in as little as 2 years. Because an OS IP PBX can be 50% lower in cost, your ROI time frame is half that of a proprietary system. It’s obviously much easier for a business to stomach a 2yr ROI than a 4-5yr ROI compared to what you would have paid for hosted services. The same reason why you wouldn’t rent a car for 2 years is exactly why you wouldn’t “rent” your IP PBX. There is still the cost of telecom services which are required with a premise based system but at least you have the flexibility of using whichever telco service you happen to find as the most cost effective and reliable- this may include POTS, T1, SIP, or some other internet based VoIP service (which I recommend against for a businesses).

I will argue that under most circumstances, a business under 25 seats will probably find more cost advantages for hosted VoIP than in purchasing an OS IP PBX simply because the investment required for a premise based IP PBX is usually more than a small business can afford. When it comes to larger organizations who focus largely on the TCO, a premise based OS IP PBX will win hands down in cost when you spread that cost out over a few years.

Hosted VoIP solutions are usually based on flat fees costing anywhere from $39 to $69 per month per station depending on the features required. Most hosted VoIP providers also require that you purchase your own IP phones which typically start at $100 for a decent device.

Hosted VoIP providers typically have a few pricing tiers which are based on features plus you may find an a la carte selection of advanced features you can purchase on a per/mth basis. Sometimes a business will be forced to purchase a higher tiered hosted VoIP product simply because they needed just 1 advanced feature which is not included in the lower tier. Some smaller hosted VoIP providers may be flexible enough to work out special pricing but this is not the norm.

OS IP PBX system on the other hand come standard with a full set of features that require no additional fee beyond the cost of the system itself. More advanced features and functionality may be an additional cost, but this cost need only be made once because there aren’t any ongoing licensing fees associated with an Open Source system. When you add a feature to a proprietary system, you may be required to pay a fee for every extension on that system, this is not the case for an OS IP PBX and therefore advanced features a far more affordable for larger systems.

Reliability:

Hosted VoIP providers treat reliability as a number 1 priority since they are responsible for the voice services of countless customers. The infrastructure on which your VoIP service operates are located in data centers with backup power, systems monitoring, high security, and much more. Since each hosted VoIP provider delivers VoIP service to thousands of customers on a single system, they can hardly afford an outage. That being said, with all the levels of redundancy in place at a hosting provider, the most common outage is typically not with the hosted VoIP infrastructure itself but with the services delivering voice calls from the hosting provider to your business location. VoIP delivery methods range from your existing internet connection, to dedicated T1’s or MetroE circuits. VoIP services delivered from most hosting providers run over a single circuit with zero redundancy. If for example, your T1 or cable goes down, so does all your voice service. This is one of the biggest problem hosting providers face when trying to deliver a reliable voice product to their clients. And because many hosting providers still rely on other Local Exchange Carriers such as Verizon for delivering these circuits, your business is now at the mercy of multiple telecommunication companies.

There is one vital flaw with hosted VoIP services and this flaw is that the majority of hosted VoIP providers rely on the internet to transmit an IP voice call. Unlike traditional phone service which utilizes a carrier’s network to send and receive calls over the PSTN, the quality of a hosted VoIP call can be largely dependent on the “weather” of the internet. Sometimes packets are sent from A to B without any loss but sometimes if the Internet is having a bad day, you’ll lose some precious data packets which will result in a choppy voice call or even worse, a dropped call. This is why most hosted VoIP providers don’t offer SLA’s since there are too many factors and external parties involved in delivering a stable voice call. This isn’t to say that VoIP over the internet doesn’t work, because it does, and it works well much of the time. It’s just not possible to guarantee the same results to every customer, and if your business needs its phone service to work 100% of the time then using an internet based hosted VoIP provider is not my recommendation.

Instead, the most reliable way to go hosted VoIP is from a carrier who owns their own network. Companies like M5 Networks will sell you a dedicated T1 over which your VoIP service is delivered. This T1 does not hit the Internet and instead carries a call from your desktop, over the T1, then directly into the carriers network, their PBX (which gives you features), and out through to the PSTN. There’s no need to hit the internet which dramatically reduces the chances of packet loss and poor voice quality.

It would be hypocritical to say that a premise based IP PBX isn’t susceptible to such service outages since often telecom services to an IP PBX are delivered over the very same T1’s, PRI’s, and MPLS circuits as hosted VoIP solutions. The reason why a premise based IP PBX can be more reliable than hosted VoIP is because a premise system can utilize redundant services such as the 100 year old technology called POTS (copper lines) which to this date is still one of the most reliable. “Should” a primary voice circuit fail, the PBX will automatically route calls over POTS thus maintaining full system functionality during a circuit outage. The cost of maintaining a few POTS for redundancy is something most small businesses can afford and certainly worth while should a primary voice circuit fail.

One great thing about hosted VoIP is that you are never responsible for its overall health and availability. Just set it and forget it! A premise based IP PBX is a piece of expensive equipment that you’re responsible for. The unique advantage of many Open Source IP PBX systems is that they’re built on the very same hardware that your IT staff are already familiar with. I would recommend that every IP PBX be supported by a vendors maintenance plan, but simple things like replacing a hardrive can be quickly done by any average IT employee.

Scalability:

Hosting providers are in the business of scaling a system infinitely. If they couldn’t, they wouldn’t be able to add additional customers. As your business grows, so can your hosted solution. What one must not forget is that for every user added to a system, there is a direct linear cost associated with the number of users added. Most OS IP PBX systems have the ability to scale significantly but instead of sharing system resources with thousands of hosted customers, a premise based IP PBX is all yours.

Hosted VoIP scales easily- just place the order and you’re good to go. This does mean that you’re entirely reliant on the schedule of your hosted VoIP provider. If you have purchased a support plan with your OS IP PBX, typically you’ll find that most vendors have a 30-60 minute response time to MAC requests which can be done remotely and usually accomplished quicker than the amount of time it would have taken with a hosted VoIP provider. There are obviously hosted VoIP providers who can be quick and IP PBX vendors that take forever so just make sure that each company has an SLA that meets your needs.

Functionality:

Functionality and features vary from hosting provider to hosting provider, but it’s safe to say that most will
deliver many of the standard features businesses require such as voice-mail, call transfer, call forward,
auto-attendant, and more. In some cases, hosted VoIP may come with very similar features to those included standard with an OS IP PBX.

The downside is that the features available with hosted VoIP solutions are limited to a finite a la carte menu. Some may come standard, where others may have an additional monthly fee, and some advanced features aren’t even an option. One of the most popular uses of VoIP for mid sized businesses is having the ability to integrate with other 3rd party internal applications for more efficient business processes. Hosted VoIP rarely has this level of versatility simply because the customer does not have the necessary access to their communications system in order to achieve integrations of this type.

Whether you need it today or not, having the ability to implement 3rd party integrations, unified communications, collaboration, call center applications, and many others is the reason why many organizations choose a premise based OS IP PBX. The Open Source nature of an IP PBX gives an organization a lot more flexibility in how quickly and easily an integration between the IP PBX and 3rd party application can be achieved.

This may change in the near future as I expect many hosted VoIP providers will begin partnering with 3rd party application developers. Already companies like Ribbit have integrated their VoIP services with Salesforce.com’s CRM package.

Tips:

The advent of VoIP as a mainstream business product has prompted thousands of hosted VoIP companies and Open Source IP PBX vendors to open up shop. I think it is almost too easy to get into the hosted VoIP business and companies like Fonality will recruit anyone as a re-seller of their phone systems. My recommendation to any small business owner is to use well established companies with existing customers and a good track record. If you’re going with a hosted VoIP provider, make sure you research whether they’ve had many network outages and for how long. Network outages happen, but they should never be frequent nor should they be for long durations….obviously. I would advise against using an internet based VoIP service if call quality is of high importance but maybe I’m just being a perfectionist since this is what most people do.

If looking at an OS IP PBX, make sure you use a company who fully understands the inner workings of the product they’re selling. Open Source systems require a more in-depth technological understanding than a simple plug and play proprietary system. Many OS IP PBX systems have easy to use admin interfaces but If you ever run into a complication with your IP PBX, you’ll be very happy to have a vendor who has the right engineering talent to solve the problem quickly.

IP external paging with Asterisk

Aaron Rosenthal — Thu, 26 Jun 2008 23:35:28 +0000

In the spirit of exploring the capabilities of Asterisk beyond its typical use as an IP PBX, external paging is a great feature to increase the value of your Asterisk system by having it assume the functionality typically reserved for a distinctly separate system and additional investment. Most people configure Asterisk systems for paging via phones but when dedicated paging is required, and when the need to page must take priority over a voice call, then external speakers are the way to go. This is often the case for hospitals and schools.

Just the other day I had a meeting with a car service organization who needs a new IP PBX system. Unfortunately they had recently made a large investment in a brand new paging system and had to postpone their IP PBX investment. When I learned that this 50 seat organization spent $30K on a dedicated paging system I whacked the guy on the back of the head for stupidity and explained that they could have purchased a brand new IP PBX + Paging system for not much more.

So some of you might ask “what external paging speakers work with Asterisk?”. Well I’ve only found one company that builds a completely open IP/SIP paging speaker and that’s CyberData. They have a whole line of paging speakers including ceiling speakers, wall speakers, voice boxes, paging amplifiers, and more…. all of which come in POE and non-POE options. Although the hardware is a little more expensive than analog paging speakers (most range from $250-$350), the fact that they can be tied into an existing investment in Asterisk and that they operate over the same ethernet wiring already in place makes the TCO very compelling.

Other than CyberData, does anyone know of or have used other SIP paging speakers?

Digium releases new AA300 Appliance to replace AA350NR

Aaron Rosenthal — Thu, 26 Jun 2008 23:34:57 +0000

Digium has just announced the release of their new AA300 Appliance. This new appliance replaces the AA350NR and have a few additional advantages over its predecessor. According to Digium, here are some of the highlights:

* More Versatile: The AA300 can be rack mounted or wall mounted. Please note that the AA300 ships with rack handles only. Rails and wall mount kits are sold separately.
* Less Expensive: The AA300 & Switchvox SMB bundle price remains the same, but the extended warranty and cold spare costs are lower.
* Capacity: The AA300 has a smaller form factor but can still handle the same number of calls and users as the AA350NR.

The AA300 will retail with the Switchvox SMB 10 user license at $4,240…. no change here over the AA350NR.
The big price change is with the cold spare which dropped from $2,295 for the AA350NR to $1,895 for a AA300NR.
As of this date, you will no longer be able to purchase the AA350.

I’m probably a bigger fan of Switchvox than any other Asterisk based IP PBX…. partially because they’re owned by Digium and partially because it’s a very solid product for the SMB market. We’ll still keep deploying our raw Asterisk systems to large enterprises but when that SMB pops by every now and again, Switchvox it is.

Botanicalls – use Asterisk to water your plants!

Aaron Rosenthal — Thu, 26 Jun 2008 23:33:40 +0000

One of the things I want to do with this blog is explore all the wild applications in which Asterisk is being used under-the-hood. Rarely do people advertise the use of Asterisk in their VoIP products so here’s one business where Asterisk was just right for a very unique application.

Although they’ve been around for over a year, Botanicalls have designed a product that lets your home plants call you whenever it gets parched or has something to say. Sounds crazy but completely true… and a brilliant use of Asterisk for a completely absurd application. By using moisture sensors that connect to the internet via a chipset and radios, Asterisk is used to dial your phone and play a message from your plant. Imagine being on vacation and getting a call.. “Hi… this is your Jade Plant calling and I’m disparately in need of a drink, please send someone over RIGHT NOW to ensure my health and safety”. The whole premise of Botanicalls is to foster the personal relationship between humans and plants by giving each plant its own unique voice and personality… keeping you in-touch with the “inner” being that is your plant/Asterisk.

I could see my grandfather getting something like this. He would go decades without a single vacation simply because he never trusted anyone but himself to determine when to water his plants/garden. Not sure how he’d feel about forming a personal relationship with his talking plants though….

What other wild and crazy Asterisk based business plans are out there?

Asterisk in the hospitality industry

Aaron Rosenthal — Thu, 26 Jun 2008 23:33:00 +0000

As more and more organizations wise up to what Asterisk could mean for their communication needs, I’ve noticed that the hospitality industry is one that is coming to surface with some unique requirements (like many industries) which I’d like to discuss and get some feedback.

First and foremost, unique billing requirements is one of the most prevalent characteristic necessary in a hospitality PBX system. Integrating with an existing billing system is usually the most common solution for satisfying these requirements but what if the customer additionally needs a complete billing solution? I’ve found that a smaller hotel won’t mind simply breaking down the necessary information from the stock excel sheets that Asterisk spits out with its CDR details. But for a larger hotel that needs a complete solution there are open source alternatives like A2Billing (not the best piece of software we’ve worked with, be careful) which is really designed for the telecom industry but can be tweaked for hospitality billing. Other OS options include AstBill and StarShop. I’d love to hear from the community what your experience with either of these two billing applications has been.

Features are another important element to a hospitality IP PBX but we all know Asterisk can satisfy almost any feature requirement. What I’m more interested in are the capabilities Asterisk can offer that may not be at the tip of your tongue. Hotels can no longer justify the expense of an IP PBX by the revenue stream generated by customers placing calls since almost everyone uses their cell phones these days. What should not be underestimated is the value-add an effective IP PBX system can have on a guests experience. Everyone still prefers to speak with a real person when at all possible, but things like scheduling a wake-up call, checking the weather, getting the hotel address, and seeing whether the pool room is open for a late night hot tub can all be automated through room phones. The key is making these features easy to access by using soft keys. You could even program the system to display the customers name while they’re checked in… imagine walking into your room and the phone is scrolling “Welcome to the Marriott Marquis Mr. & Mrs. Jones”. I’d give my hotel major points for that!

The next and last component of a hospitality PBX that I’d like to discuss are the phones. Phones are what every patron sees sitting on their bed-side table and as higher-end hotels continue deploying new electronic devices in their rooms, phones should not be overlooked (but I think quite often they are). Sure most people won’t use their phone to make personal calls but a room phone should display relevant information on its screen. By using a phone with a large and sexy color display, each guest room phone can serve up information like the weather, scrolling stock quotes, and even advertising. Advertising is an excellent way to justify the additional expense of purchasing a color phone. Advertising this way can be an excellent revenue stream.

Teledex, a manufacturer of phones specifically for the hospitality industry, released last year a line of IP/SIP phones. Although we’ve never used these phones with Asterisk, Teledex claims that they work well with Asterisk. Now I don’t know how Teledex has avoided the Apple Inc. hammer but their IP phones are named the iPhone series. Perhaps the best feature on these phones is that they use a touch-screen color display. Watch the Teledex iPhone demo and you’ll get an idea of how awesome a properly configured IP PBX system can be a huge value-add for guests. The color Cisco 7970 is another nice color phone which is used by the hospitality industry and I’m hoping that the new Polycom IP670 with its color screen will also be a great guest room phone.

You don’t hear much about Asterisk being used by large hotel chains but hopefully this changes and I know we’ve been seeing many more opportunities in this industry over last year. In addition to what’s already been discussed here, what other features and functionality has this community seen or worked on for the hospitality industry?

Get to know SIP if you haven’t already

Aaron Rosenthal — Thu, 26 Jun 2008 23:30:39 +0000

I still find it surprising the number of IT directors and CIO’s who still have no idea what SIP is. This is mostly the case with SMB businesses but even if you’re a small multi-site organization, you’re not doing anyone justice by ignoring new technologies. Because I run into so many people who don’t have a solid grasp of what SIP can do, I wanted to write a post laying out some examples.

So what is SIP? Only the greatest communications protocol in the world! Go wiki SIP for a technical overview but I’d like to discuss the benefits of SIP by using a simple example. Consultants and sales people don’t necessarily need to know how specifically SIP works but they should really understand what it could do for their clients. I find that people are less aware of SIP simply because carriers are slow to roll out their own solid SIP services since doing so can cannibalize some of their own business.

There are two primary methods for delivering SIP to a customer. The first is via the Internet and the other is via a dedicated circuit from the phone company. For a mid sized business I always recommend dedicated circuits because it’s the only way I can offer an SLA backed quality guarantee since there are far too many factors out of my control to guarantee voice quality over the Internet. This SIP example is a perfect demonstration of how SIP and MPLS can help dramatically improve the communications of a mid-sized organization with 3 locations.

I have a little medical center that operates 3 equally sized offices with approx. 50 employees at each site. Before we came into the picture, each site had its own independent legacy Avaya system connected to its own dedicated voice T1. Each site also had a dedicated data T1. Because the EMR system was located at the HQ site, each medical center was additionally connected to the main HQ via a point-to-point for application sharing. So in total this medical center was paying for 3 voice T1’s, 3 data T1’s, 2 point-to-points, and supporting 3 disparate analog Avaya systems. This setup is absolutely absurd from today’s standards but 8 years ago it would have been completely normal.

What was the solution? Simple as this: Replace all 3 Avaya systems with a single Asterisk system at HQ, deliver a 10meg MetroE to each of the 3 sites thus replacing all T1’s and point-to-points….and that’s it! By using SIP, we can route all voice traffic from the 2 sites through HQ then out to the PSTN. Data was dramatically increased from 1.5meg to 10meg and now all three sites can call one another for free while being able to transfer calls between sites, all the while having a fully meshed data network for their application sharing (made possible by MPLS). This is a HUGE improvement from the old setup… the best part being a projected savings of ~$4K/month which pays for the new phone system in 2.5 years.

Talk about a perfect client needing a complete communications overhaul. The encouraging part is that there are still thousands of businesses out there with the same horrible setup… we just have to find them and whip them into shape!

I recommend that you check out the SIP Forum to learn more about SIP and see what else other telecom professionals are doing.

Deploying a phone system is one thing… getting users comfortable is another…

Aaron Rosenthal — Thu, 26 Jun 2008 01:44:20 +0000

One of the biggest hurdles a company faces when deploying new technologies is the re-training of an entire user population. This is a much larger task when implementing something like a new ERP or even a CRM solution but one must not underestimate that users can also be slow to learning new phones.

Normally a phone will come with a user manual that explains everything but who has time to read a full phone manual??? I recently came across a company called VoIPTrainer.com and they have some interesting web based videos designed specifically for the user who just had a new phone thrown on their new desk. VoIP Trainers videos are easy to follow and offer just the right information a typical phone user would need to figure out their new phones.

I really wish that every phone manufacturer like Cisco, Nortel, Avaya, Grandstream, Polycom, Snom, Aastra, and everyone else, would create their own web based user manual just like these by VoIPTrainer… It would make so much sense! VoIPTrainer has videos for most of the Cisco 7900 series which I like, too bad they don’t have training videos for any other products. They’re obviously a new company but I think they’re onto a good thing, now they just have to do my favorite phones, Polycom!

For those of you who use Cisco phones with their Asterisk systems, this might be a nice value-add or tool for helping users familiarize themselves with their new desk accessory. And until they come out with Polycom training videos, I’m still stuck with my own condensed version of the Polycom manual… shucks.