The Cruft Of My Brain

Well, I guess I better get a post in for the month of November. Ugh, it’s been a long month. Anyway, fiddling with free podcasting tools and methods and I discovered some updates in Snow Leopard that are pretty cool.

I little while ago on Leopard I found out that with iLife 09 you could use GarageBand to record a voice chat using Bonjour in iChat. That’s cool but kind of limited. I’m not certain but I think this was not extended to non-Bonjour voice chat’s. That’s not the case with Snow Leopard. Not only can you record your Jabber and GTalk sessions but it’s smart enough to create the right number of tracks if you have a group chat going on with multiple users. And yes, it’ll adjust the title picture track based on the users avatar from GTalk for whoever’s talking.

I’m a little shocked that it works as well as it does.

One thing I haven’t been able to figure out is how to start recording a voice chat and then add in additional local mic’s. It doesn’t seem to be possible so all recording must be done with the group voice chat participants. And I haven’t tried this but I’m guessing that adding users to the group voice chat will not start new tracks in GarageBand. That would be slick if it did.

There’s been a lot of noise lately about AT&T and Apple and Google not playing nice together. The latest is AT&T carrying on about Google Voice blocking rural numbers because their costs are so much higher. Ok, that’s kind of crappy, but it’s just a red herring. AT&T is making a fuss over this because they also see the real potential. What happens when Google Voice links up with the Google Talk service. I’ve been thinking about this for a while and wondering why GTalk wasn’t a supported endpoint for GVoice yet. I thought it might be technical but now I have my doubts.

At this point GVoice still relies on the telco’s to provide the actual phone service. If GTalk becomes an endpoint then there’s a much stronger argument for Google being a “carrier” instead of just an “internet service”. That brings all of the issues of access, fees and taxes along with it. Would Google have to start collecting a USF and other similar fees for every GTalk user, even pure IM users, on the basis they “could” use the voice component? Pricey! As far as I know, every other bridge to the PSTN (SIP gateways mostly) has to pay the fees and support universal access.

I think that’s the shot across the bow from AT&T and it’s something Google’s been aware of for a while. And it upsets me greatly.

The world is headed in the direction of more XMPP and SIP, not less. AT&T sees this and probably sees Google as the biggest target to slow this adoption down. AT&T is using the admittedly noble idea of universal communication access to beat down it’s competitor. The FCC should instead be looking at how to address issues like the USF in a world of online focused communications. The trend for land line terminations is only going to increase to the point where only poor people have landlines, subsidized by the USF. Where will those fees come from when everyone is online focused? This isn’t a new thing and it’s obviously been one of the top issues for the FCC.

Perhaps the USF should be added to (more) data lines. Or the Fed Gov can just take it from us with other taxes. They seem to be pretty good at that already.

Most of all, it’s shameful that AT&T is using USF and universal access as a hammer to beat on it’s competitor. Hey AT&T, how about being more competitive and innovative instead? I won’t hold my breath.

I can’t say I blame them. I’m sure there are some very qualified programmers at Arc that are proud of their product. My experience was less than stellar.

So, here’s what I’ll offer. It seems there might be a new version out. I’ll try a demo copy on a fresh WinXP install if they will provide it to me. I’m not going to drop the $1000+ they ask for this software (there’s another criticism) to see if times have changed. If one of the recent Arc commenters want’s to hook me up with a demo copy though, I’ll test it. I have a clean CCME I can point it at for testing as well.

I’ll post my results, good or bad. If the new version really is so much better I’ll happily post a comment to the original blog entry that seems to have climbed onto the first page at Google. (I’m guessing that’s why this is getting noticed)

So, what do you say Arc? Leave me a comment!

Bennie from Arc found my blog and wrote a couple of comments on this post. I started writing a response but it got a bit lengthy for a comment so I decided to throw it into a new post:

Bennie, thanks for responding. That install was almost half a year ago. Ancient history!

Regarding my comments, I hate Java. Arc Express, Cisco’s ASDM, Cisco’s abomination for router configuration (can’t remember the name now) all use Java and to varying degrees show off what’s horrible about it. My overriding problem with Java is that it breaks User Interface conventions. Or rather, it has it’s own set of them and developers implement them inconsistently. The result is often a mess of a program where users have to struggle with silly things like inconsistent window focus and “enter” not doing anything. It’s 2009, I shouldn’t be forced to click the ok button because someone couldn’t program what the “enter” key should do in a text field.

It’s a whole slew of oddities like this that Arc Express seems to have hit on, all combining for a frustrating experience. Regarding this experience I seem to have purged the whole thing from my memory. You can see from my comments that I had numerous problems with UI elements acting inconsistently.

Do you really want my suggestion? Run away from Java. Drop it like it’s hot. Also, your product has all of the hallmarks of being written by engineers, for engineers. Maybe you’re ok with that but I don’t think that’s the market you’re going for.

Lastly, my memory is a little fuzzy on this but I seem to recall having to jump through some strange hoops to get the license or access to the install guide. I don’t recall exactly what the deal was with that but I remember it being bizarre and frustrating. It’s just an attendant app but somehow it’s like you guys want to build a certified installer ecosystem. Maybe I’m remembering it incorrectly. It’s just an app though. Sorry to burst your bubble but you don’t warrant an entire certified installer ecosystem.

As an alternative, look at Asterisk’s Flash Operator Panel (FOP). This program runs on the server, is included with just about every Asterisk distribution and for the most part just works. Now, I’m not real thrilled about Flash as that has potential to be another rathole like Java. Still, they’ll probably be able to port their app to HTML5 pretty soon.

In a nutshell, don’t “be different” when it comes to UI because the programmer thinks it’s cool. Especially because the programmer thinks it’s cool. That and accept that although you guys have a deal with Cisco, that doesn’t mean you are Cisco.

Sorry, one last thing. The UI I’m referring to here is almost universally the install UI, not the actual attendant console. As I referenced before I think the attendant console is actually ok. Not great but ok. This is one area where you have non-technical end users and they probably would benefit from thinking differently about the UI. One feature I love on my iPhone is conferencing multiple parties together. It’s painfully obvious how it works and it works great!

I’m not bringing this up because it’s something new…but rather because it took Cisco ages to add this to their least expensive voice solution, the CME. I have an inbound SIP trunk set up with Vitelity for my business. It comes right into the same system that’s running the home phones. It’s so nice to be able to direct that inbound route based on the DID I get from Vitelity. From there it’s a pretty simple matter to send the call where I want. Right now I direct it at a RingGroup which does RingAll. I then have my office phone and my softphone ring for business calls.

At the same time I don’t have to change the house side of things. I also have a RingAll RingGroup for the home POTS line. The POTS comes in through a Cisco 1760 which is then plar’ed to my RingGroup extension via SIP. I have the same office phone in that RingGroup as well. I prepend either Vz or Anewvo (company name) to the CID and I can see where the call is originating from. That prepending is done either in the inbound router or the RingGroup. Flexibility!

Next up, I think I’ll try some “follow me” on the business line. It’s all in there, should be pretty easy to configure.

On a related note, I’ve been quite happy with the call quality on the Vitelity trunk. It’s not like I’m hammering my link when I’m on calls but so far it’s been excellent. Price is pretty reasonable too!

I keep an eye on my logs and for as long as I can remember this post has been the most popular:  http://www.thecruftofmybrain.com/2006/03/14/cisco-router-with-fxo-as-an-asterisk-gateway/

It’s funny that almost 3 years later very little has changed.  I’m running Trixbox CE now but the configuration is still pretty much the same.  I’m pretty happy with how TB has progressed and I’ve stuck with it.  For a while I was running it virtually and while that worked ok it always had problems with the clock slipping.  I’m now running on an older P4 and all’s well.

Regarding my old config, a couple of things seem to come up with the folks that find that post.  First, you don’t want the OPX version of PLAR.  I’m not really sure why Asterisk isn’t happy with it but it has to do with the order of going off-hook.  Just use the regular PLAR and you’ll be fine.  Also, make sure you put the trunk into the proper context on Asterisk.  I say “proper” because everyone will be different.

If someone has a good resource for understanding the proper way to configure contexts, please let me know.  I understand the basic nature of it, after all it’s similar to Partitions and Calling Search Spaces on CallManager.  But it doesn’t seem clear to me that there’s a simple way to manage them.

It’s been a while since I looked at what was formerly known as the Cisco Attendant Console. Color me unimpressed. I found the new version’s install process to be tedious mainly because it’s java based. Key buttons would disappear repeatedly during the install and I had to reboot several times. The original promise of java creating this coding utopia has been completely corrupted by the large volume of incompatibilities between java versions.

Once I got it installed it seemed to work ok, after yet one more reboot.

The funny thing is that Arc is one of those weird companies that likes to do things differently. They just seem to make things more difficult than it needs to be.

But hey, at the end of the day they’re the only game in town so what can you do. I guess it’s nice to be a monopoly.

Gizmo5 is finally the one to break the Skype barrier.  They have a new service now called OpenSky (har) that is essentially a proxy between Gizmo’s SIP network and the Skype network.  I imagine they’re doing something like the non-free Skype Asterisk channels that have been out for a while.  This is cool but it would have been even more interesting when Skype was offering free out-calling.  Now, I’m not sure how big a deal it is.

Ultimately, the SIP world needs to come together, perhaps with XMPP as the discovery mechanism, in order to support universal and open calling.  I’ve been fiddling with IM on domains for Google and it works great!  More of that is needed!

And yes, I’m aware of ENUM but that doesn’t seem to be making progress in North America.  Someone show me that I’m wrong about that.

Once again, I’ve moved the in house phone system.  I was running into problems with the virtual version I had running for the last couple of months.  It worked pretty well most of the time but the clock was continually out of sync and that caused other problems on occasion.  We had one instance of the server rebooting in the middle of a call.  Not cool.  Good thing it’s just at home.

I’m sure there’s a way to make it work with the proper VMWare tools and a good bit of tweaking but it’s just not worth my “time”!

So, I downloaded the freshest copies of TrixBox CE and chan_sccp_b.  Again, I followed the steps here and had no problem getting everything installed.  Remember to change the IP address for the server in the xml files in the tftpboot directory and also in the sccp.conf file in /etc/asterisk/ and then reboot the whole thing.  Reloading Asterisk didn’t seem to be enough.  Not sure why that is.  Restart your phones and you’re good to go.  Oh yeah, in my case I had to change the SIP related IP’s on my 1760 gateway also.

After all of the struggles in the past with the sccp channel it’s amazing to me how far it’s come.  It really is pretty easy now.  It sure would be nice if they could build a GUI module for the configuration though.  In time perhaps.

Part of my motivation to do this is to fix the clock problem, but I also plan on getting a little more complex by adding some VoIP based connections to the world.  So far I’ve only had the single Verizon analog line.  I think I’m going to start by getting an inbound number from Gizmo5.  They’ve built in integration with Gizmo5 now and it’s pretty easy to set up.  I’ve been testing the outbound calls since they give you $2.00 in free calls at 1.9 cents/minute for signing up.  The quality’s been excellent!  The only problem has been that my parents don’t answer because they don’t recognize the number.    Buying the inbound number should fix that.

What I really need is a small form factor TrixBox CE server.  Something along the lines of what Digium is offering in their Asterisk Power Appliance.  Something cheap that can handle <10 phones would be a real sweet spot.

It’s official, I’m going independent.  I know, I probably could have picked better market timing.  I think we’ll be getting a bump in the next month or so though and this year might turn out ok.  Time will tell.

In the meantime, I need to find some contract work!  I have a number of opportunities lined up but the process of lining up more work needs to start now.

If you haven’t guessed from reading this blog, I’m primarily a Cisco guy focused on Core, Voice and Wireless.  I’ve done projects as large as managing a team of engineers (and being the lead engineer) on a 30+ 6500 switch rollout (both hybrid and native mixed) to installing and fixing single T1 scale routers.  A fair bit of the work I’ve done in the last few years has involved either voice or wireless.  I’ve designed and assisted with the rollout of a number of multisite Cisco Voice solutions.  I’ve performed wireless site surveys and then designed wireless solutions based on the results.  Those wireless solutions have generally been LWAPP based which I also implemented.  In the course of performing these wireless implementations I’ve converted standalone APs to LWAPP, configured Guest access with the portal and installed coverage maps in WCS.

I currently hold a CCNP and CCDA and I’m pursuing a CCVP.  I have a number of partner focused certs such as:

• Express Foundation Design Specialist
• Unity Design Specialist
• Rich Media Communications Specialist.

I previously held the Wireless SE and FE certs but have not renewed them due to role requirements within Cisco’s partner certification process.

So, I’m looking for interesting contract based gigs in the Metro DC area.  Baltimore is great too.

More to come as things develop!