Skype for Asterisk: Lowering the Cost of Business Communication

For those of who have never heard of Skype, it’s a service that allows users to make free calls from their computers over the internet to other Skype users; and ultra-low cost calls to landlines and mobile phones globally.  Asterisk is a popular open source PBX system that manages the routing of calls in addition to other features including unified communications.  Companies like Dalcon provide the software that makes it easy to put this valuable new communications tool to use.

The Advantages of Skype and Asterisk

The main advantage of Skype and Asterisk is of course, value.  Both Skype and Asterisk have been well known in the VoIP world as low cost alternatives to more traditional forms of communication.  The growths of these applications have been spectacular, and as more businesses learn that the value they bring doesn’t come with a sacrifice in quality, the more they will continue to grow.

Skype to Asterisk, using software like the Dalcon Communications Manager, allows your company to:

1. Make free calls globally from any phone on the Asterisk PBX to Skype users.
2. Make low cost domestic and international calls (either with unlimited minutes subscriptions or low costs such as 2.1 cents per minute to many global destinations).
3. Allow Skype users to call you on a Skype user address, receive the call on your Asterisk PBX and then route the call to any extension , IVR/auto attendant, queue, or even outside cell phone.
4. Place a Skype button on your website so Skype users can call your company instantly from their computer.
5. Allow Skype users to make a free Skype call from anywhere in the world into a conference bridge on the Asterisk system so that you can have a mix of Skype, internal, and external phone users on one conference call.
6. Purchase a Skype-In landline phone number for your company ($5/month) in most any city worldwide. Calls to that number can ring into your company’s Asterisk/DCM system and be routed in any way you choose.

How it works

Once you have an Asterisk-based PBX, such as the DCM, you can order Skype for Asterisk connectors and sign up for your Skype business account.

Now your options are limitless.  You can make Skype calls using softphones and desktop phones. Asterisk can be configured in many different ways to automatically switch between Skype calls and normal calls.  For example, you might set the system up so that dialing *9 on a desktop phone allows you to “SkypeOut”—in other words, make a Skype call out to an international landline or mobile phone.  Another possibility could be simply configuring Asterisk to automatically SkypeOut when you dial a number with an international country code.  Using DCM you can set up any Skype user as an extension on your system and when you dial that extension on your desk phone it automatically routes the call to the Skype user over the Skype-to-Skype network.

Asterisk Guide: How to Install Asterisk (Part 2 of 2)

Last week we discussed some of the less effective routes for Asterisk help and documentation.  This week, I’d like to direct you to some of the more efficient ways to attain information on the subject.

An Asterisk Reference Model

“So what am I supposed to do” you ask?  I’ve found that the developers at Digium working so diligently on the Asterisk code base are actually very good at keeping documentation.  What I didn’t know, was that they only seem to keep that documentation in the source releases.  To make sure you’re doing things correctly for YOUR version (I’m assuming 1.6+ but this should suffice for other versions as well) do the following :

1. Download the source tarball for YOUR version
2. Uncompress it
3. Go to the “configs” directory
4. Read every single file here
5. Eat some pie (This is an important step)

This accomplishes three things:

•    You now have a sample of *nearly* every config file that Asterisk needs to run.  These can be dropped into your /etc/asterisk directory as they are, and it will help you go a LONG way in getting things setup.
•    These files are immensely commented with helpful hints and usage examples for all of the things you need.
•    You have a belly full of pie.
“But I don’t WANT to comb through all these configuration files to try and figure out what I need and what I don’t.  Can you help me cheat a little?”  Some files are definitely more important than others.  Pay special attention to the following:

•    extensions.conf – The brain of your PBX.  This determines how calls are routed.
•    sip.conf – This is where you configure your phones
•    voicemail.conf – Pretty obvious
•    meetme.conf – Virtual conference rooms
•    queues.conf – Call queues

The rest of the files really depend on a couple of variables.  First, it’s important to know the basic pieces involved in setting up a fully functional PBX.

•    The Asterisk source code
•    Zaptel or DAHDI source that matches your Asterisk version
- Asterisk 1.4.22+ will use DAHDI instead of Zaptel
•    If you’re using them, the drivers for your telephony gateway device (PRI card, analog card)
- It should be noted that Zaptel/DAHDI contain the drivers for any Digium based cards.

Zaptel and DAHDI have their own configuration files and you’ll need to understand them as well.  Typically, there are helper tools designed to create these files for you, but that often depends on the type of hardware and drivers that you’re using.

The Zaptel files are:
•    /etc/zaptel.conf
•    /etc/asterisk/zapata.conf

While the latest versions (1.4.22+) will use the DAHDI naming scheme instead:
•    /etc/dahdi/system.conf
•    /etc/asterisk/chan_dahdi.conf

That should go a long way in getting you started.  I realize I didn’t give specifics (as I said I wouldn’t be earlier) but in my opinion, just diving into a working system will still leave you feeling overwhelmed when it comes time to actually configure a dialplan.  If a straight walkthrough is what you want, those are available via Google.

Of course, if all of this seems too daunting a task for the time you have, there are companies who specialize in Asterisk based phone systems.  Using Asterisk allows them (and us) to provide a lower cost solution with a lot of the same bells and whistles as the larger vendors.

Michael McNeil has been working with Voice Over IP technologies since February 2006 with Dalcon Communication Systems. He specializes in Asterisk: The Open Source PBX, Linux, and Perl development with a short background in Network Security.

Asterisk Guide: How to Install Asterisk (Part 1 of 2)

Whether it’s technical curiosity, or a low-cost business need, a lot of you (if you’re reading this article) have researched or played around with Asterisk: The Open Source PBX. It’s almost too good to be true, right? What’s not to love about a free open source software phone system your company can use that is also highly configurable by yours truly? In this article, I hope to help shed a bit of light on the pros and cons of working with Asterisk.

Let me start out by saying that you can install Asterisk pretty easily if you’re looking for something simple. However, with around 30 potentially large configuration files that need some form of special care, and add to it the hassle with compiling the appropriate packages and drivers to make your telephony hardware function, that walk through the park can quickly turn into a winding maze with strange growling noises from within. It’s not a pretty place to be, especially if you’re spending paid work hours to dedicate yourself to the task.

A Quick Start

First of all, let’s look at a few ways you can speed the process up:

•  AsteriskNow – (http://www.asterisknow.org/)
•  Trixbox – (http://www.trixbox.org)
•  Debian Linux – (http://www.voip-info.org/wiki/view/Running+Asterisk+on+Debian)

The above approaches are perfectly fine for average scenarios, but nearly every company/person out there has some sort of special need. Typically it’s a feature available in the latest version of Asterisk, but not currently available in your drop-in product.

I won’t detail any of the major operating system debacles you may come to face. Seriously, that would just take way too much time. I also won’t be covering specifics on compilation. Again, there are too many paths for you to take and it’s unrealistic to think I could cover them all. I just want to try and help you find the quickest answers you need without wasting your time.

Important Resources

Here are some of the avenues I’ve used to get where I am today:

•  Google – (http://google.com) – This is nearly every tech’s best friend
•  Voip-Info – (http://voip-info.org) – A site dedicated to Voice Over IP with huge influence from Asterisk.
•  IRC – (http://java.freenode.net) – You’ll have to register a nickname here and confirm it via the email they send you. Then you can /join #asterisk.

In all honesty, these are the only three tools I ever needed in order to make my way in the Asterisk telephony world at first. Over time, however, I found that voip-info.org didn’t always specify which versions of Asterisk a certain feature was available in. The information there usually provided several different ways to do something, and while some may work in your version the others would crash Asterisk!

IRC is really one of the best places to get the answers you need when troubled with a problem in your Asterisk installation. Sadly, however, there only ever seemed to be a handful of folks who are able (or willing) to give suggestions without a full listing of every configuration file you possess. I certainly understand the argument; Asterisk is an immensely configurable piece of software that can produce widely varying results if you tweak a setting in the wrong way. Still, I despise having to post every single configuration file on the web just so I can wait an hour to find that they couldn’t figure it out either. This probably should go without saying, but I will say it anyway, that going into the IRC channel for help without having some semblance of a working Asterisk installation is a bad idea.

Check in with us next week, and we’ll discuss some better avenues of help for your Asterisk related endeavors.

Michael McNeil has been working with Voice Over IP technologies since February 2006 with Dalcon Communication Systems. He specializes in Asterisk: The Open Source PBX, Linux, and Perl development with a short background in Network Security.

It’s Time to Switch to SIP Trunking

By now most people responsible for their company’s phone system have heard about SIP trunking as an alternative to using a PRI circuit to connect to the PSTN. Many have also heard about SIP Trunking’s advantages over PRI, and that it is “the future of telecom.” But often they wonder when it is appropriate to take the plunge and switch to SIP?

First we’ll cover some of the basic advantages of SIP Trunking over PRI’s as well as some things to watch out for when considering SIP Trunking; then we’ll cover some potential opportunities to switch to SIP.

Advantages of SIP Trunking over PRI

The main advantages of SIP Trunking over PRI are Cost savings and scalability; let’s list a couple:

1. Less equipment costs: Equipment costs for VoIP systems that use SIP Trunking instead of PRI are between $2000 and $6000 less per IP PBX.
2. Smaller Monthly Fees: Each SIP Trunk voiceline costs between $15-25 per month, while each PRI voiceline typically costs between $30-$65 per month.
3. Smaller Calling Fees: Calls with SIP Trunk phone systems have extremely low costs compared to PRI calls. Outbound call savings alone could quickly cover the costs of your new phone system if your organization makes a lot of outbound calls.
4. Scalability: SIP Trunk voicelines can be added 1 at a time, with PRI you are limited to adding groups of 23 voicelines at a time.

The potential “gotchas” with SIP Trunking are related to how you are connected to the SIP Trunking provider. DSL and Cable connections can be too inconsistent for acceptable quality voice, especially if multiple simultaneous calls are needed. How many “hops” are involved between your PBX and the SIP carrier’s point of presence can also be a factor in the controllable quality of the voice.

Opportunities to Switch to SIP

SIP Trunking is clearly where telecom is headed, but when is it appropriate to make the switch? Here are some great opportunities for your company to switch over to SIP Trunking:

•  Your telecom contract is coming up for renewal: Most telecom service contracts are written for 1-3 yrs. When they are up for renewal is the best time to look at a switch to SIP Trunking. CAUTION—Many contracts are written for automatic renewal if not terminated a specified time before contract term end.
•  Switching from TDM to VoIP: By now most businesses are using VoIP to communicate, but if you’re one of the few remaining TDM users, now is the time to make the switch to VoIP. Because SIP Trunking requires less equipment than PRI, upgrading to VoIP with SIP Trunking costs about the same as putting in a new “old” TDM system.
•  Business Growth: Is your business growing? Business growth is a great time to switch to SIP Trunking because you can add additional voicelines 1 at a time instead of in groups of 23.
•  High Calling Costs: If your business makes a lot of outbound calls, switching to SIP Trunking will dramatically lower your monthly phone bill.
•  Multiple Business Locations: If your business has multiple locations, they can communicate SIP-to-SIP and completely bypass the PSTN—resulting in calls between business sites that cost fractions of a penny.
•  Right Now: With the advantages SIP Trunking provides over PRI, any time is a good time to inquire about SIP Trunking.

In summary, SIP Trunking is a proven solution to lowering your telecom costs. So the next time you receive your monthly bill consider making the switch to SIP Trunking.

Dalcon Communication Systems provides telephony solutions. Our initial consultation is free and there is no commitment—so give us a call at (877)WE-UNIFY to see if we can help you start saving money on your telecommunications today.