gundy dot org

I’m happy to learn that Asterisk 1.4 is out.

The new version of Asterisk (1.4) is getting a GUI. I’ve known that for a while. I didn’t know, however, that Digium has wrapped the whole thing up as a Distro based on what looks like Fedora Core or RHEL.

From the announcement:

Asterisk® can now be easily configured with a graphical interface. The new site, AsteriskNOW.org, which is still in development, hosts AsteriskNOW™ Beta. AsteriskNOW™ Beta is a Software Appliance; a GUI implementation with the open source Asterisk distribution. AsteriskNOW includes all the Linux components necessary to run, debug and build Asterisk, and only those components, so installation is easy. You no longer have to worry about kernel versions and package dependencies. Unlike other Linux distributions used to deploy Asterisk, no unnecessary components that might compromise security or performance are included.

I’m usually very suspicious of GUIs used to configure servers, but if Digium is behind it, I’ll at least give it a look. A “distro” based install of Asterisk sounds like a great trouble-free approach when installing for someone that doesn’t have full-time sysadmins.

I’m still going to be using CentOS or RHEL for my personal stuff and wondering about the real question… When will the Zaptel drivers make it into the kernel?

Update: Looks like the distro is rPath Linux.

Things are always harder then they’re supposed to be. Plugging in a SIP client to Asterisk should be easy, right? Well, it is, if everything is just perfect. I took a few notes and a few screenshots to help others if they find this fairly typical setup is more difficult then it should be.

Here are the relevant parts (IP addresses and mailbox number changed to protect the innocent) of my sip.conf:

[general]
port => 5060
bindaddr => aaa.bbb.ccc.ddd
disallow => all
allow => ulaw
allow => alaw
context => sip-users
canreinvite => no
nat => no

[gabe]
username => gabe
secret => ****
mailbox => 1337
callerid => Gabe Gunderson <1337>
type => friend
host => dynamic

To to avoid NAT issues (on my side, not the server side), this communication is going over an OpenVPN connection. Make sure your tunnel is up before you start Ekiga and go to the preferences page. Set “Listen On” to tun0 and not eth0 (which is the default). The settings don’t seem to “take” until you restart Ekiga.

I like to keep things simple in the codec department. The VSP we’re using offers only G.711 (U-LAW or PCMU and A-LAW or PCMA). We’re using U-LAW since it’s the standard here in the U.S. and it costs the CPU very little to go from U-LAW < => U-LAW. It’s a little fatter in the tubes, but that’s not a big deal on this particular install.

Here’s a typical Account. I’ve chosen to use names for sip users instead of numbers. It makes the configs a little easier to read and you can always map numbers to names in the dial-plan.

If you have problems, there are a few things you can do to help figure out what’s going on.
Try the following:

• Run Ekiga from the command line with the debug switch:
  ‘ekiga -d 2′
• Or get specific debugging from Asterisk, like so:
  server*CLI> sip debug peer gabe
  SIP Debugging Enabled for IP: aaa.bbb.ccc.ddd:5060

If this needs explained, you won’t think it’s cool anyway ;)

vi /etc/asterisk/extensions.conf
;; Call hacks.
[hacks]
exten => 411,1,Dial(${TRUNK}/18003733411)

It makes it so that when you dial 411 (directory service) your call actually goes to a free 411 service. Myself, I have a hard time remembering long toll-free numbers. This makes it easy and free!

This blog details a new service that allows you to make FREE phone calls… there are strings attached.

The deal is that you get a free 6 minute phone call, but you give up your privacy. The calls get published on the Internet for the world to hear. It’s better then tapping the pay phone at the state liquor store at 2:00 A.M. And so, we now get to sit and listen as people “air their dirty laundry” to the world for the price of a 25 cent phone call. Jerry Springer has nothing on these guys.

From the source blog:

He (James Bowman of for60secs.com) explained that they envision a few different kinds of user for the phone service, such as people who are very price-conscious or shameless self-promoters who view it like reality TV - you give up privacy in exchange for fame.

And now you will never have to wonder “So she is 100% alive?” Warning: many of the calls contain strong language.

How did this information cross my path? It’s powered by the very cool Open Source Asterisk PBX.

Ah VoIP…

My caller ID shows up as:

“COMFORT INN SUI” <3038045380>

7 out of 10 is good if your are playing “name that tune”. For caller ID, it might as well be 0 out of 10. The good news is that my VTP is better to deal with then Qwest.

Oh yeah, and my name is not Mr. Comfort Sui.

Here’s a pretty good write-up on how you can use Jingle with Asterisk for “calling” over Google Talk.
This is cool stuff.

• You could have calls coming into work (Asterisk server) ring your office phone twice and then try your office and Google Talk client simultaneously.
• You could get a call at home that your wife answers. She knows you’re not home, but sees that you are on IM. She could transfer the call to your laptop and the caller would have no idea that you are in a hotel in Las Vegas.
• You could put all your geeky buddies with Google Talk accounts on speed-dial.
• You could place calls from anywhere you have an Internet connection and route them through your Asterisk server at home.
• You could have Asterisk IM you with the caller ID of the person calling you.

Did you notice all of those points say “you could”? I have not tried this yet, but when I do, I’ll report.

Look for full Jingle support in Asterisk 1.4. Until then, you need this patch.

P.S. I wrote about Asterisk/Jingle/Google Talk in this earlier post when I first learned that it was being worked on.

Looks like Digium gets funded. This can only mean good things for the development of Asterisk. It’s interesting to see that Digium has been in the black for a while:

Digium won’t disclose any specific data on its financial performance, but says it has been profitable since 2002, generating 100 percent growth in revenues each year since.

It takes one kind of vision to create a business around Open Source, but another to make that same business something *big*. Hopefully this first round of funding leads to the next big thing and makes a good thing even better.

However, the cynic in me doesn’t really care how they do. The code will always be there - in good times and bad. Is that wrong?

O.K. I’m over it. Go Digium! Support Digium! Buy Digium!

The humble server that I use to host a few (11) web sites is getting updated. It’s not going to be that big of a deal (99% idle instead of 97% idle).

The big deal will be the addition of Xen, LVM and RAID. I’m thinking about setting up a small domU that will act as a firewall. It will also run Apache as a reverse proxy and Postfix as a transport relay to the domains that will get their own DomU (3 currently). Normally I would also have it run Asterisk to avoid NATing with SIP, but since I switched to IAX with my voice provider, I don’t think it will be necessary.

Anyway, the lights might go out for a little while. Don’t be alarmed.

I’ve been busy this weekend hacking together a little something I call ZoolTone.

What is it you ask?

It’s a Firefox extension (my first, but not last) that will allow you to “Click and Dial” phone numbers on a web page. It adds a pretty little icon after every phone number it finds. When you click the icon, it sends a message to Asterisk to make the call. Asterisk then rings your phone with the caller id of the number you clicked. When you answer, it connects you to the other party. It should be pretty sweet for offices that run Asterisk.

It’s called ZoolTone because it’s written with JavaScript and XUL which is pronounced “zool.” I kept all the logic in the backend to make it easy to add Thunderbird support later. It would be so sweet to be reading a business email and call that person simply by clicking on the phone number at the bottom of page.

With real work starting on Monday, this will get neglected for a bit. I do hope to have a working version out in a week or so. Any longer would kill me.