What’s IT all about?

I have recently upgraded my mobile phone to a Nokia E51, partly because I wanted to test out it’s potential as a SIP extension on our Asterisk network. This, of course, opens up the possibility of having one phone that you use whether you are in the office, on the road, or even at home. The detail of these scenarios needs to be determined by your particular circumstances, but it is certainly feasible.

There is plenty of information on how to actually set up this phone on an Asterisk system (Google is your friend here), so I will merely say that this aspect of the test was pretty painless.  Within minutes of taking the E51 out of it’s box I had it registered as an extension and happily making and receiving calls. Personally I would rather not have this as my only office phone, so any incoming calls are actually routed to both my desk phone and my E51, but you may wish to set it up differently.

Whilst the E51 has great battery life in what might be described as a traditional setup (i.e. not using the wi-fi capability) it is a known fact that wi-fi does drain the battery much more quickly than would otherwise be the case. This phone is very much business-oriented, so is designed to have pretty good battery life (amazing when you have it in your hand and feel how slim it is). I have heard estimates of 4 or 5 days between charges mentioned. All that goes out of the window when you turn wi-fi on. I have been doing a lot of testing over the last few days, so my use is atypical, but I have still had to charge the phone once a day ! I would expect that to improve slightly when my testing is complete, but I can’t imagine it lasting more than 2 days.

Anyway, back to the Asterisk interoperability. The E51 allows you to choose which type of call routing (i.e. mobile or internet) to use as default. If you choose internet, it will always fall back on standard mobile calling should the internet route fail. In practice this seems to work well, with one exception.

Having one device for all situations means having a single telephone address book. At first glance this might seem to be ‘a good thing’ and in truth it is…but it did throw up one interesting problem. As I travel abroad from time to time I like to store phone numbers in my mobile in ‘international’ forma, namely with a plus sign at the start, followed by the 2-digit country code, followed by the actual number. So, for instance, I would store our office number as +441233888240

This is all well and dandy, except the first time I fed such a number into an internet call, and therefore on to Asterisk, it didn’t work. It didn’t work to the extent that the call was not made rather than passing through from Asterisk to the mobile network. Thinking the problem was in Asterisk, I looked in the log and found….nothing ! A bit more testing suggested the phone was trying to place the call via Asterisk, but it wasn’t getting anywhere.

To cut a long story short, it transpires that passing a plus sign to Asterisk, without specific consideration for that within an outbound route, will result in the call being dropped pretty unceremoniously. So I proceeded to add the following to the outbound route dial pattern ‘+|.’ This matches the plus sign, strips it out and then passes the rest of the string on to the relevant trunk. This is important as stripping more out would remove the country code, and you’d get into a bit of a tizz trying to put it back on.

The IAX2 trunk I use for most outbound calls expects national numbers without country codes or international numbers starting with a ‘0′ followed by the country code. Since I wasn’t passing numbers in either of these formats, and didn’t want to reconstruct my dialing logic completely, I set up a second trunk in asterisk for the same provider. I haven’t tested this with a SIP provider, and I have my suspicions that it wouldn’t work, but for IAX2 it is fine. Within the second trunk I know what’s coming in (i.e. an international number starting with the 2-digit country code) so I merely tag ‘00′ on to the start  (using ‘00+Z.’) and send it off down the trunk. So far it works very smoothly and I can have all my mobile phonebook numbers in international format.

Yesterday I received in the post some information from a company (who shall remain unnamed) in the public sector who had put out an Invitation To Tender (ITT) for a new, VoIP-enabled telephone system. The ITT itself contained very little information indeed about their requirements, as is often the case. However, it did mention the fact that it was a public sector company, and since value for money is frequently highest on their selection criteria, Asterisk-based systems are usually a good fit. So I cheerfully applied for an information pack, which duly arrived, as I said, yesterday.

A quick read through the detailed tender document made it apparent that this was one contract not worth taking any further. When you see selection criteria such as

“The system will not be based on open source code”

then you suspect that they have had their fingers burnt. A more in depth perusal confirmed the suspicion:

“Since implementation London users and Homeworkers have experienced a variety of
issues with voice quality. The voice quality has varied from acceptable to unusable.
Outbound voice quality has been mostly acceptable but inbound voice quality has on
occasions contained dropouts or has been completely inaudible, preventing users
from hearing what callers are saying.”

They go on to say that WAN changes and the introduction of G.729 have improved matters, but obviously not enough. They also bemoan the lack of available expert Asterisk knowledge making it difficult to troubleshoot effectively.

There are a number of checks that any half-decent Asterisk consultant should have made at this site. Is the server up to the job? Is there adequate bandwidth for their requirements? Has QoS been implemented correctly across the board? Are there unnecessary codec translation going on? Is the VoIP provider up to the job? Instead company or consultant who carried out the initial implementation has left them in the lurch, and now a company that could have been very happy with an Asterisk solution are determined never to touch it again, and if anyone asks their opinion they will no doubt be very disparaging.

The solution? The best way to avoid the situation, in my opinion, is for any consultant or company selling an Asterisk-based solution/service to make sure that their customers have an alternative contact should things go belly-up. Do your homework and get in touch with someone who knows their way around Asterisk blindfolded. Your customer is more likely, not less, to stick with you if you can demonstrate that you have their best interests at heart and can provide them with a level of service that they require. That does not need you to be an Asterisk guru, but you do need to know where to turn if such a person is required.

Just noticed that Atcom have a new firmware available for the AT-530 (v070813). Right now there does not seem to be any information on their website about what issues it is supposed to address, or what features it adds (nothing I can see). However, in the interests of my loyal readers, I decided to install it anyway and check it out for you.

As far as I can see, there are no new features, although a previously reported issue with the Daylight Savings setting does now appear to have been resolved. However, the update was smooth and the phone appears to have suffered no ill effects. So I cautiously recommend you give it a go, having ensured you have a backup of your current config first, of course.

If you spot any changes that I haven’t, then please let me know. I will keep an eye on the Atcom website and update this post if any information appears.

Update: A missed call that resulted in voicemail on my Asterisk server has also resulted in ‘New Messages’ being displayed on the phone display, and the status light is also flashing. That’s the good news, the bad news is that listening to, and even deleting, the voicemail has not removed the message or flashing light. Haven’t yet found any other way of clearing the new found MWI functionality either. Ho hum !

An interesting news item on the Teleappliant site here.

Of course, Teleappliant aren’t the only company to offer an Asterisk-based PBX

One sometimes overlooked advantage of using VoIP technology over traditional telephony solutions is the enablement of remote access to the phone system. So rather than having to physically be at your desk (or at least in the office if you use a DECT phone), you can now have your work phone number follow you around wherever you have internet access. Many corporations, and a number of smaller businesses, who have jumped on the VoIP bandwagon have enabled this aspect of the technology already, and find it extremely useful. There is one important consideration, though, that you should make if you are considering this course of action yourself.

In the VoIP world, your voice traffic follows a very similar route to your data traffic. In smaller companies especially, the temptation therefore is to utilise the existing data network infrastructure to ease the implementation of the voice network. However, since the voice traffic needs to be routed via the internet, you end up compromising your edge-of-network security to implement VoIP (for instance, the recommendation for RTP traffic is to open UDP ports 10000 to 20000 !). Bigger companies will probably separate voice and data network equipment as much as they can to mitigate this risk, but smaller companies may not wish to, for financial or other reasons. Introduce the desire to allow remote soft or hard phones to login to your company PBX so that calls to their DID can follow them around the world, and you can start to see the extent of the risk.

The solution ? Well, as in many cases, that depends on the company and how much effort they are willing to put into identifying and addressing the risks. The only real solution is to run a proper risk assessment exercise so that you understand what could happen, the likelihood of it happening, and what you have to do to fix it.

The result, though, is peace of mind to go with your mobile VoIP telephony.

The world of VoIP promises a lot, and not the least of those promises is free phone calls via the internet. The reality, at least for now, is not quite that simple; but then is anything ever that simple ? That’s not to say that free internet calls are not possible, rather that you have to meet certain criteria before you can reach that exalted goal. In essence, free calls are an option if you and the person you are calling are on the same network. So if both of you are using Skype, or are Vonage customers, or use any other internet telephony provider that you should be able to chat away for hours without any problems or charges. The issues arise when you both are on different networks.

Skype

First of all, let’s take Skype as an example. One of the barriers to non-Skype users is that Skype uses a proprietary protocol, and as barriers go that’s a good ‘un ! Therefore, if you’re using a SIP-compliant PBX/provide, the world of Skype users is pretty much closed to you unless, of course, you have Skype running alongside your SIP softphone/handset. Not ideal, but about as good as it gets. (Note: there is a product called PSGw which will integrate Skype and SIP, but it requires Skype to be running on your PC so I’m not convinced you gain an awful lot.)

SIP

If you are on a SIP network (there are many providers such as Vonage, Sipgate, Voiptalk, FreeWorldDialup etc.) then you have a few more options. Most SIP ITSPs (Internet Telephone Service Providers) will have peering arrangements with other SIP ITSPs which allow you to route calls directly from one network to another. Therefore, if you are a Sipgate customer and you are calling a FreeWorldDialup customer then, with the addition of **777 at the front of their phone number, your call does not touch the PSTN and you incur no charges. Nice, but with a couple of limitations:

1. Limited peering. You are reliant on your ITSP having peering arrangements with many other ITSPs, and some are better than others in this respect.
2. Foreknowledge. It only works if you know beforehand which ITSP your contact uses, and then go to the trouble of programming in the right prefix for them. So for new or rarely used contacts you will invariably end up just using the standard number. (Note: SIP addresses go part-way to fixing this, so that if you publish your SIP address (e.g. SIP: 5576167@sipgate.co.uk) on business cards, emails, etc. then that information can be used to place free calls as long as appropriate peering is in place.)

Asterisk

The addition of your own Asterisk server (or any other PBX for that matter) into the mix opens up another option. Now, instead of having your softphone or shiny new SIP (or even IAX2) handset hooked up to your ITSPs PBX, you can run your own PBX, and only route calls through your ITSP when necessary. The advantage of this kind of setup is that you can accept incoming calls that are not routed through your ITSP, and this is important. Why, because you now have a means of avoiding the ITSP peering restrictions that would otherwise restrict from whom you can accept SIP calls. You are essentially setting yourself up as your own ITSP and can decide with whom you wish to peer.

“Hang on” you say, “doesn’t the other person need to know your server details in order to call you for free?”. And yes, you are quite right. As with peering, if the calling party doesn’t know you can accept a call directly then the default route is through the PSTN and back out through your ITSP. This is the issue that SIP addresses are supposed to resolve at the ITSP level (see above), and the mechanism for resolving this problem when you are running your own PBX is ENUM.

ENUM is most easily described as DNS for telephone numbers. For instance, on the e164.org site you can register your VoIP phone numbers and add ENUM records that will translate that number into an IP address, domain name or even a SIP address. It’s similar to publicising a SIP address, except in theory it is much more powerful for the following reasons:

1. Set and forget. Rather than relying on the person calling you to have received your SIP address information somehow, you are now relying on them performing an ENUM lookup. This is a one time operation for the PBX owner, making it much more likely to be done.
2. It’s all about control. And in particular, putting you in control. You set up your ENUM records, and you can change them should your circumstances change.

ENUM is more likely to be used in a business environment as only a small percentage of SOHO VoIP users will run their own PBX, so it is certainly still sensible to publish a SIP address for now. For home users, ENUM lookup on outgoing calls is a feature offered by some ITSP’s, although given that it will only ever reduce their income, you can understand why many are not offering it. For them there is a balance to be found between making money from SIP to PSTN calls, and improving their attractiveness to potential subscribers by having a lot of peering arrangements or ENUM lookups. Right now there are relatively few VoIP numbers out there…but that will only grow.

Conclusion

So, to summarise, it is possible to maximise the number of free VoIP calls you can make, but you just need to do a little groundwork first.

• Check your contacts. If most of your potential VoIP-enabled contacts are Skype users, then use Skype to call them. If they have SIP facilities, then implement a SIP solution.
• Start small. The world of VoIP is still low on the growth curve and thus is changing constantly. So right now it makes little sense (unless you’re a large business) to implement a costly solution. Keep it simple for now.
• Plan for growth. Having said “Keep it simple”, you should also factor in likely growth. Not so applicable for domestic solutions, but for a growing business you might want to think twice about putting a Skype handset on every desk.
• Ask for advice. There are a lot of open source solutions (such as Asterisk) in the VoIP arena, and many people willing to offer advice. Be wary of the guys who shoehorn their ‘product’ into any scenario you can come up with, but keep an open mind.

OK, shameless plug time. If you are a growing business looking for help and advice on a VoIP solution, then please feel free to email or call. PSTN number is (UK) 01233 888240.

Rather than give you chapter and verse, have a read of this article on Digium, the company behind the Asterisk Open Source PBX and you might get an inkling as to the opportunities still available to you when you can’t charge for licences.

My review of the AT-530 did not go unnoticed, particularly one of my critical comments regarding the display which was permanently set to ‘VOIP Phone’ - not particularly useful I’m sure you’ll agree. Well, a couple of days ago an email arrived with a rather large attachment - a new firmware update hot off the Atcom presses (now also available on the Atcom website download area).

The new firmware offers a few new features:

Version 07-04-24 Release Note:

　　1.Add eight different ring tone
　　2.synchronize from SNPT server
　　3.Add the sliding bar of menu setting page
　　4.Add LCD logo display customization function
　　5.Display the current server type ( SIP or IAX)

You’ll note No. 4, which I can confirm works. I have yet to check the other new features (keep an eye out for the full review soon), but this new firmware does appear to have broken one previously working function. Now, if you set the time zone to GMT then you just cannot enable daylight savings, so the clock will be an hour slow during the summer months. Back to the drawing board guys !

I have been having problems getting emailing of voicemail to work on my home Asterisk system….seems it’s due to the lack of an internal DNS server ! This article explains how to fix the problem.

I’ve been thinking about a hypothetical discussion. This chat is with the owner of a smallish company who is running an old PBX which, despite a number of frustrations, has served them pretty reliably over the years. The conversation may be face-to-face or on the phone; it may be in their office, our office or at an exhibition. Wherever it is, there is one simple assumption about how this conversation would come about that I should share; namely that there is some, as yet unknown, reason why this person is considering replacing their phone system. So the first challenge in any such conversation is working out why that person is talking to you.

Notice that I’m considering the question from the customer’s viewpoint. A common mistake I’ve seen over the years (and made myself, if I’m honest) is to take the wrong viewpoint. It’s like a carpet salesman saying to a potential customer “I think you should have this carpet because it’s easy for us to lay.” It’s not going to impress the customer. In fact you may end up without a customer.

So the business owner is standing/sitting in front of me and we have a chat about why they’re there, what’s happening to the business over the coming weeks and months, how they are getting on with their current PBX, etc. I would expect this conversation to serve up the information I’m looking for. So I may find out they are attracted by the cheaper calls offered by VoIP, they may be relocating to bigger offices, they may find that the running costs of their current system are rising as they need an engineer in to make even the most trivial change, they may even be reaching the end of the maintenance/service agreement on their current system and it’s a natural time to consider other options.

There may even be other reasons that I haven’t considered (answers on a postcard….or at least in the comments). Once I know the driver, I can start explaining how our product is so much better But that’s for another blog.