1. Cisco’s brand has a better reputation. The Cisco name means one thing: networking.  When your name is identified with your core competency more than any other name, it means you make good products, and have done so for a long time.  Avaya, on the other hand, has a name that is new since the late nineties or early 2000s, a name that was cooked up as a way of shaking the legacy image of Lucent/AT&T.  But, as the old saying goes, if you put crap in a pretty box with a bow, it’s still crap.   Of course we know that Avaya’s products aren’t crap, but, in hindsight, and setting aside trademark law for a moment, would it not have been better to stick with the name Lucent?

2. Cisco has a more qualified sales channel. Cisco’s resellers are data people, many of whom have also been supporting software applications for decades.  Avaya’s resellers are “phone guys”.  Two entirely different kinds of folks. Software people understand the needs of users.  Phone guys often don’t.  The reason is thus: software can be made to do whatever the user wants, however telephone systems, a la TDM, always had a rigid feature-set that didn’t evolve around innovative user ideas.  So Cisco resellers, having sold and supported software for a long time along Cisco’s gear, are often more experienced at tailoring solutions.

3. Cisco spends more on product placement than Avaya does on payroll.  OK, maybe not THAT much, but you get the picture. Everywhere you look, Cisco’s products are on the screen and in print.  If you were an alien from outer space watching American TV or movies for the first time, you would think that every phone in America was a Cisco 7970.

4. Cisco, with the Bells’ help, convinced the public that SIP is some kind of new, experimental, risky protocol. Of course, nothing could be less true.  But, since SIP holds the key to creating interoperability between all UC telephony platforms, Cisco has been in no hurry to hype it.  Not to mention all the licensing fees they’d be giving back if their customers began to abandon the Skinny protocol.  Avaya, now’s your chance.  Hype SIP’s scalability and global compatibility before Cisco grabs another 15% of your market share and converts those endpoints to SCCP.

5. Cisco has succeeded in defining the “standard features” of a VoIP phone setup. This despite the fact that those features aren’t all that different from a traditional TDM setup, and despite the fact that there are actually fewer features and less flexibility in CallManager than there is in Avaya’s solution.  Avaya, if you want to win this skirmish, open up your media server and let users create their own features–not by selling them expensive call-center add-ons (that would be Cisco’s approach), but by just including EVERY piece of call-center software in EVERY distribution.  You’ll be surprised how quickly customers flock to you.  Remember that you’re locking customers in to a 12 – 15 year platform investment.  If you GIVE them the flexibility and power that Cisco will only SELL them, they’ll come.

SwitchVox is a San Diego-based IP Telephony technology integrator that has been producing very eyeball-friendly Asterisk phone systems for the last several years. Brian and Tristan Degenhardt, both with SwitchVox since its early days, were instrumental in publishing my second book, VoIP Hacks, as Brian contributed some fantastic material for the book, and Tristan coordinated his contributions.

I met Tristan briefly at Fall VON Boston two years ago (in the Asterisk Pavilion), but didn’t have much time to go over things with her.  Well, as it turns out, SwitchVox was very soon later acquired by Digium, and the rest, as they say, is history.

The AA60 is aimed at small offices (there are more capable models aimed at larger offices, of course). Street price on these is around $3400 for a 10-user bundle.  Now I’ve never been a big fan of licensing users for PBX access (especially on an open-source system), but I realize there’s no better way of appropriately monetizing the offering.

Initial setup of the AA60 involves connecting a PC keyboard, mouse, and monitor to the rear panel of the AA60 (shown a few scrolls down).  Once you’ve done the network setup, the keyboard, mouse, and monitor are no longer needed, and the unit is configured through a fantastic web interface. Actually, SwitchVox’s web interface is arguably the main reason Digium acquired the company.

So the AA60 is really a PC. It does have sort of an odd form-factor. I was expecting it to be 19″ rack-mount standard, but its enclosure is about the size of a slimline/SFF desktop PC, maybe a bit bigger.  Digium does include a mounting bracket for placing the AA60 on a wall board.

Now the AA60 doesn’t come with any legacy telephony interfacing out of the box, though Digium’s cards can be configured and installed as a part of your order. This would allow you to equip the unit with T1/E1 PRI access or analog trunk/station ability.  I have a Digium Wildcard TDM with two stations and two trunks that I’m going to try out as a part of this demonstration, so I’ll let you know how that goes.

The AA60, unlike other VoIP appliances (such as the Jazinga we looked at a few weeks back), delivers only voice functionality. That is, it isn’t a switch, router, or firewall.  For installers looking at a more high-end PBX product with fewer strings attached, this is a blessing.  For some offices, and all-in-one unit makes sense. I would argue though, that for most, having a standalone PBX makes the most sense.  Keeping PBX separate from infrastructure spreads out the points of failure and doesn’t make your phone system reliant on a bundled switch or router. Digium has wisely decided not to include those extra components.

In the next post, we’ll get into configuring the AA60 and talk about the pros and cons, if we discover any. See you then.

And, I was going to videotape the whole process so I could share the ups and downs with folks on YouTube. I plugged the Linksys phone into the LAN and went to get my camcorder.  But, by the time I got back with it, which was about 2 minutes, the phone was ALREADY RUNNING on the Jazinga system. So it went from out the factory box to being a working SIP peer on the Jazinga system, firmware config and all, in under two minutes, and the best part–I did NOTHING, just as Jazinga claimed. Heck, I didn’t even have to key the MAC address of the phone into the Jazinga box.

Those clever Canadians are pretty good at this VoIP stuff–they should keep it up!

Jazinga is a Canadian VoIP software startup that offers a small business telephone solution, among other VoIP technology procurements. The solution works with SIP phones and consists of a shiny grey appliance that certainly conjures thoughts of its contemporary, the Asterisk Appliance, another VoIP telephone system from Digium. When I first looked at the Asterisk Appliance, I was mostly impressed–there were a few key things missing, though, and if Jazinga can seize on Digium’s shortcomings with their product, I’ll give ‘em a high-five.

First off, know that the Jazinga product is not yet available to the general public, however the software and hardware I looked at was in a very functional state. So it’s fair to compare with the Digium device, since it too was in a pre-release state when I looked at it.

Let’s talk about the Asterisk Appliance for a minute. It looks great, offers support for up to four analog phone lines, and has a “pretty good” GUI web interface–nothing as slick as Switchvox or Fonality, but functional.

The Asterisk Appliance MUST be directly connected to the Internet in order to act as a VoIP trunk gateway, and this is hugely bad. For one, the Linux under the hood of the Asterisk box isn’t all that easy to keep up-to-date security patch wise, and for two, the underlying routing, NAT, and access functions necessary to the Appliance’s functioning as a router aren’t available via its web interface. That’s why I haven’t actually used an Asterisk appliance in the field yet.

Small businesses don’t have a block of IP addresses to work with, and typically don’t have a DMZ segment to sit their phone system on, so I haven’t been able to recommend it as a production PBX.

Let’s hope Jazinga has solved some of these problems.

The first thing I noticed about the Jazinga box was its inclusion of an antenna, ostensibly making it a wireless access point as well as a PBX. A quick glance at the back panel of the main unit, and I realized it’s also an Ethernet switch and a broadband router, just like the Digium unit. The Jazinga offers 4 LAN ports, one WAN port, two analog phone connections, and one analog phone line connection.

According to the quick-start card Jazinga includes, the Jazinga must act as the Internet gateway, which is somewhat problematic. Like the Digium unit, which frustrates me that it has to be the Internet gateway router, the Jazinga insists on being “directly” connected to the Internet–ie. there cannot be a NAT point in front of the Jazinga as it communicates with hosts on the Internet. This is unfortunately, if excusable. VoIP signaling protocols still haven’t totally bridged the NAT gap, but the Jazinga innovates its way out of that mess by allowing you to disable routing altogether, and by providing support for IAX trunks–which are immune to NAT problems normally imposed on VoIP protocols.

Initial setup was easy. I only had two snags. The documentation I received said to use a URL with a host of jazinga.local, but the DNS obtained from the Jazinga through DHCP didn’t resolve that name. Fortunately a quick ifconfig told me what IP address the Jazinga was on and it was mostly smooth sailing from there. The second issue I ran into was a difficulty with the DHCP client on the WAN side. In this case, I was using the Jazinga with a cable modem, which, after a reboot, got the Time Warner DHCP server back on track. So, pretty easy.

Jazinga is going to offer a branded VoIP trunk service especially for this device. At this point I don’t know where their PRIs sit, but I expect you’ll be able to get a local number anywhere in N. America. So, not wanting to be held up, I got to work connecting the Jazinga to my Gizmo account, with which I have a Gizmo CallIn number.

I’ll post the results of that experiment and the rest of my Jazinga evaluation in a day or two. Stay tuned.