One of the first GUIs for Asterisk was Asterisk @Home which eventually became Trixbox . Now there are tens of them. On the voip-info page I’ve managed to count about more than 40. Some are completely free, some commercial of which some  also have free counterparts.  Many offer payed support, training and certification.

So how do you choose between 40+ choices?

It’s easy you don’t. One should never have more than 3 things to choose from.

Sow how do you choose, really?

Well I guess it all comes down to which of the GUI distributions or front-ends you came in contact first. Because you have the most time invested in that one.  Or you trusted someone’s recommendation. You cannot possibly make an informed choice, on your own, by trying and testing 46 distors.

Upgrading

There’s another issue, the upgrade. What happens when you want the latest Asterisk version on your distro? If you have a commercial one the vendor will eventually do it for you, but that might cost you both time and money. If not, than you’re at the mercy of the community maintaining the distro.

Maybe you’re involved,.

Maybe you should be, it’s your business.

Extending

What about extendability? Your clients want some functionality that’s not in your GUI of choice? You will definitely have to do it yourself.   How many files have to be changed, additional “things” installed, changes to the database. It’s pretty easy while you’re doing it the first time, and fun too, but how do you repeat the process?   Add that same “module” to some other installation?  Document everything? Yeah that’s the fun part. And do the same 37 things all over again.

No wait I’ll make a script. And then, one day, the script  stops working. They changed the distro, or whatever. So now you have script maintenance issues. Two clients with version 0.9.8.4.6.1.3.ra and one with version  0.9.8.7.4.8.6.rb and who knows how many others with random strings of characters in version numbers. Is it really worth it? I’ll just say that we don’t offer that functionality any more.

Installation

What about the initial installation ?

Easy, intuitive? Easy is not a bad word. Easy should be a prerogative. Doesn’t mean that you’re smart if you’re doing something a rookie is unable to do. It only means that you have time to waist. Asterisk is easy to install. Why? Because it’s predictable and because someone (Digium) keeps the order of things.

Compatibility

And what happens when you want to add some exotic interface to your PBX.

Well if it’s plain Asterisk no problem. Once I needed to change the misdn channel’s source files, but that was just one file. I saved the diff and that was a no-brainer to repeat. Later Digium updated the files so that was not necessary to do any more.

But with a big GUI distro, that made a mess of Astrisk’s cont files, that becomes an order of magnitude more complicated.

Windows infamous DLL Hell is nothing when compared to all of the above.

Support

And what if you’re doing five other things? At the moment I’m doing at least 7. Besides the Asterisk maintenance and integration, there’s coding, web design, SEO, what not, accounting, business stuff… Do I really need to “learn” another paradigm? Of a questionable benefit and quality. It’s not like I’m trying to comprehend the string theory. It’s a mishmash of 20 -30 code stacks glued together in an unnatural liaison. And all the effort goes into making sure that the parts do not not fall out of the stack. So that brings us to the support. You need some way of keeping up with things.

The commercial GUIs offer payed support, training and certification. I would only pay for Digium’s excellent boot camp and dCAP (I plan to, in the future).

So what do I suggest? API. There is an API, the manager.

The Experience

Many may disagree with me, but all those GUIs are uuuugly. That may be due to not being from the same code base. I know that many people from the Asterisk community think that to be a non-issue, but just look at Apple. What do you think sold the first one million  ipods?. It was the beautiful user experience. Beautiful as in looks and beautiful as in simplicity.

Spying on me

It seems that some of GUI implementations have a “phone home” mechanism.

This is true for TrixBox and FreePBX. You can check out this post from TrixBox forums or this one from SlashDot. What if you’re doing a PBX for a government agency. In that case you (the PBX integrator) are committing a felony!

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I’ve been using Asterisk for years and never really wanted to use Skype, but since many of my friends do, I decided to have a go. It was quite unthinkable to use Skype separately from Asterisk, I mean I have all of my communication unified through Asterisk box, and now this thing called Skype had to be exempted from the rest. No way. So I was looking for a way to connect them. Skype uses proprietary protocol for communication, and the company DOES NOT ALLOW reverse engineering. Therefore, there was no “clean” way of making a Skype channel for Asterisk, like it was done with Cisco’s Skinny protocol. Well, how do we connect then?

There are two possible approaches: hardware and software one. Hardware approach would require some “Skype to phone” adapter “thingy” that connects to Windows PC via USB, and has FXO and FXS, through which it then connects to Asterisk via Sipura 3000, or what have you. I excluded the “enterprise level multi-channel skype gateways” because they are too expensive. I mentioned Windows PC because I had one up and running 24/7, as described in the previous post.

Available software

I decided upon software approach, since I did have available Windows PC, and didn’t want additional “Skype thingy” and addional Sipura. Here are my choices:

• ChanSkype, Asterisk channel driver for Skype. It sounds great but has one major flaw: it requires X server and I want my Asterisk box to run clean without any additional burdening services.
• PSGx, Personal Skype to H.323/SIP gateway. I have been using it for a while, and it turned out to be buggy and unreliable.
• Uplink Skype to SIP adapter is what I am using right now.

Windows installation

You can download Uplink Skype to SIP here.

When you run the setup program you get a dialog like this:

Then you just let Windows install the unsigned drivers. These audio drivers are the heart of the program. The program sets up these drivers to work as Skype’s headset, and then translates the sound coming from them into SIP packets:

As shown in the figure  (Skype audio settings), the sound coming from you, through the microphone, is getting into SIP packets via Uplink’s sound driver, and vice versa for the sound of the speaker. There are similar hacks with the actual microphone and speakers, that is hardware ones, to find out more check out Grynx’s blog.

Uplink settings

Account settings:

• SIP number and password : your Skype number/nickname and password that are the same as in sip.conf.

• Server: Asterisk IP address.

Network settings:

• Listening local port: default for these settings is not the same as Asterisk’s default SIP and RTP settings, so pay attention.
• Public IP address of your Asterisk box..
• External SIP and external RTP also differ from Asterisk’s default settings.

Asterisk Skype2SIP configuration

here’s my configuration:

sip.conf

[tokenskype]
type=friend
host=dynamic
nat=yes
username=tokenskype
secret=tokenskype
canreinvite=no
insecure=very
dtmfmode=inband
disallow=all
allow=alaw
context=incoming_skype
callgroup=2
pickupgroup=2
canreinvite=yes
qualify=yes

extensions.conf

exten => 711,1,SetCallerID(${MY_ID})
exten => 711,n,Dial(SIP/echo123@tokenskype,40,rwW)
exten => 711,n,Hangup()

Here you see extension for Skype’s echo test. Buddies should be added the same way.

Putting it all to work

After everything is installed and upon starting Skype2SIP you should see something like this:

And if you do “sip show peers” in Asterisk’s prompt:

Calling test echo extension:

and Uplink’s screen shows (talking for 52 sec, then hanging up, from 9:41 to 10:33):

Now just add a few buddies, give them extension numbers, and you’re set to talk…

Just a few final words:

• I made a context for incoming Skype calls “incoming_skype”, as you see in sip.conf excerpt . There you can customize it to dial a specific phone, softphone, send e-mail or whatever…
• Skype video calls can not be routed through Asterisk, but Asterisk is capable of conducting video calls, which is an entirely new topic…

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