Job Postings

If you are a corporate client looking for someone to work on a PBX for you in the cheapest and most efficient manner, you have come to the right place. Take a look at our forum and post a job offer that our freelancing Asterisk-ers can take a look at. It’s easy, free, and confidential.

On the other hand, if you are a freelancing VoIP telephony specialist, you may want to post an excerpt of your resume in our freelancer section of the forum. As said before, it’s easy, free, and totally confidential!

So before you sit down and worry about cold calling those 100 businesses that you had in mind, why not publish your resume or job offer here and let us do all the hard work for you?

New Asterisk Blog Forums

After putting many hours of work into setting up the forums, they are finally ready to be released.

If you have been looking for a better way to discuss Asterisk or VoIP in general, then this is the way to go. You will get a response quickly and accurately for any questions that you may have. Make sure to pop over and register so that you can be a fully involved part of the community.

As thanks for being such loyal readers, the first 50 member to the forums will recieve a free voicemail on my private toll free Asterisk box as well as free custom @AsteriskBlog.com email! Just sign up and send me a private message (send it to user “chris) and I will get it set up ASAP!

IAX Provider Reviews

I am in the market for a new provider (as we all know, NuFone is not exactly the best around), but I’m stuck looking for a new company. That is why I need your help!

I’m opening up the website for a little discussion. Those of you who have an IAX provider that you either own or swear by, leave a comment linking to the website along with a short review. The sooner I get a new provider, the sooner we can have a dedicated line for this website with all sorts of cool features!

Nerd Vittles: Tricking out your TrixBox

Nerd Vittles posted today about some awesome tips and tricks and a couple of fallbacks on the new TrixBox 1.2 release. Make sure to check out the full article.

View the full article

For those that thought we’d dropped off the face of the planet, good news. Not yet. If you haven’t heard, there’s a new version of TrixBox, 1.2. And we’ve given it the old college try for a week or two with about that same results pictured in this old comic book. On some platforms, it runs just fine. On others, including our VMware for Windows machines, it’s a nightmare. The voice synthesis system is again broken, freePBX can’t reload Asterisk without completely shutting down and restarting Asterisk (amportal restart). And there appear to be all sorts of interrupt or timing problems that we’ve never seen before … going back to Asterisk@Home 1.2. We attribute many of the problems to a new version of CentOS and Asterisk, both of which are bundled into the TrixBox 1.2 package, but who knows. What we do know is TrixBox 1.2 is a little too Bleeding Edge for our taste, and most of the Nerd Vittles goodies that depend upon the Flite speech engine no longer work on many machines.

Trixbox Review – Part 1

I downloaded Trixbox today and installed it so I could begin the review of it. Let me first say that this was by far the easiest distro to install.

Trixbox runs on CentOS 4.4. The AAH distro’s ran on CentOS as well, but I never had any luck at getting those to install. Trixbox was a snap. I threw in the cd, chose a couple of options, such as my time zone, keyboard type, and so on, and twenty minutes later, it was installed. Trixbox did an excellent job of autodetecting things.

I was half expecting a desktop though. I was thinking it would be very similar to Asterisk@Home. I was wrong. When it booted into the prompt, it kind of surprised me. Not a big deal though. It’s a server, so I don’t really need much in the way of a desktop anyway, I am just fine with that.

If you choose to install it, after your first bootup, type in “trixbox-help” and it will bring up an options menu. It has about 20 different reccomendations of things to do. Go through them and choose what you will.

Basically so far I have just installed it. I have played a little bit with FreePBX, but nothing too extensive. I’ll write more of a review once I have it up and running. Overall though, I give the installation a 5 out of 5!

Rss Feeds

Hey guys! Great news. I added a new section to the site, which you can view by clicking “Rss Feeds” above. It is a section of VoIP related feeds that I am syndicating here. Make sure you check them out and read some of the great information held within. Thanks everyone!

Sniffin’ the VOIP traffic

This time we will install a network protocol analyzer to watch the traffic on our LAN from initiating and connecting a SIP call.

The Wireshark open source project was formerly known as Ethereal. I used to work for a great company called Cybera as a programmer, and I was always fascinated by networking. I’d bug the network engineers for any information I could, and play around with Ethereal to try to understand what they were talking about.

If you’re working under windows, download the installer. For our Ubuntu or Debian friends, it’s available under the standard free apt archives.

There’s one little trick you need to be aware of during the install.

winpcap

Make sure you select WinPCAP as part of the installed goods.Complete the install and start the program. Minimize it for the time being.

Launch your VMWare server and the Trixbox instance, log in, and you’ll notice the IP address shown after you log in. Mine is 192.163.1.93.

Run over to another box on your LAN and make sure you can ping this address, as detailed in my last post.

If you don’t see ‘Logged In’ in the faux LCD window, most likely you’ll need to update the IP address that the phone needs for Asterisk.

Click the little Menu button juuuust to the left of the green phone button. Select System Settings->Sip Proxy->Default.

Menu

Make sure that the IP address for Domain/Realm, SIP Proxy, and Outbound proxy are all set to the IP address of the Asterisk Trixbox server you just started via VMWare.

Remeber, Nerd Vittles set us up with 500 and 501 as 2 extensions to use with these phones. Dial 501 from the 500 phone or vice versa. I launched mine just now and I can hear the kids, dog, and my wife doing fun stuff. Frankly at this point I have to sit back and marvel at the processes running to make this possible. It just blows my mind.
Now comes the hackin’ part. As the SIP call is in progress, flip back to Wireshark.

wiresharc-startup.PNG

From the main window, select Capture->Interfaces.

wriesharcints.PNG

I can see one of the listed network interfaces dealing with a lot of traffic. Choose that one and press the capture button.

wriesharkcaping.PNG

Let wireshark capture at least 5 or so seconds of traffic. So far, on mine, the vast majority of this VOIP traffic is of the UDP variety. Click Stop and wireshark will dump it all into its analysis window.

analyze.PNG

Every line that says OICQ Protocol represents one UDP (User Datagram Protocol) VOIP packet traversing the network. As a side note, it appears that Wireshark has made the assumption for us that these packets are really part of a chat protocol popular in China, which, of course, is not correct.

Right click on one, and select ‘Open in new window’. Go down to the bottom and look at the ‘data’ section of the packet. This data section represents the actual digitized voice of the VOIP call. It’s interesting to me that the protocol used is UDP, which is one of the two major types of IP packets, the other being TCP. UDP is a connectionless protocol, which means that the client generating the traffic simply puts the packet on the wire without regard to checking to see if the recipient actually received it. This also implies that the recipient has to collect the correct UDP packets and reorder them to form a meaningful conversation. I wonder what role the SIP ‘stack’ in asterisk plays in this function. I suppose we’ll find out here at Asteriskblog!

Well, I hope you’ve found that illuminating, and I’m sure we’ll be referring to this tool to diagnose our further work in Asterisk. Please contact me if you have any questions.