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Some good internal news at TMC. Technology Marketing Corporation today announced that Digium CEO Danny Windham has accepted its invitation to deliver a Keynote Address at ITEXPO East 2010, taking place Jan. 20-22, 2010 at the Miami Beach Convention Center in Miami, Florida.

Windham's presentation, which takes place Thursday, Jan. 21 9:45 a.m. EST, will address the evolution of Open Source as a mature telephony platform that is experiencing extensive growth in enterprise, government and carrier markets. Open to all ITEXPO attendees, this session will also be a centerpiece of the fourth annual Digium|Asterisk World conference, which is collocated with ITEXPO East 2010.

ITEXPO is the world's largest conference and trade show focused on communications and technology. Launched in 1999, the event is expected to draw as many as 200 exhibiting companies and more than 7,000 attendees to Miami.

"Developers, resellers, IT professionals and executives will benefit from hearing Danny's views on the increasing number of opportunities based on Asterisk, the world's most popular open source telephony software," said Rich Tehrani, CEO and group editor-in-chief of TMC. "ITEXPO and Digium|Asterisk World will expand on this notion, with three days of content, education and training for anyone interested in learning more about open source telephony."

Digium is the creator, sponsor and primary developer of Asterisk. Digium offers Asterisk software free to the open source community and offers Asterisk Business Edition and Switchvox IP PBX software to power a broad family of products for small, medium and large businesses. The company's product line includes a wide range of hardware and software to enable resellers and customers to implement turnkey VoIP systems or to design their own custom telephony solutions.

Windham joined Digium in February 2007 as CEO. He is responsible for setting the company's corporate strategy, and executing its day to day business operations. Prior to joining Digium, Windham served as president and chief operating officer of ADTRAN, a global provider of networking and communications equipment. He joined ADTRAN in 1989, following ADTRAN's successful acquisition of Processing Telecom Technologies, a company Windham co-founded in 1986. Windham holds a Bachelor of Science degree in Electrical Engineering from Mississippi State University, where he was named a Distinguished Engineering Fellow in 2001 and he also holds an MBA from the Florida Institute of Technology.

"Whether building a corporate phone system from the ground up, or extending the functionality of a legacy telephony system, organizations of all sizes are increasingly turning to open source VoIP as a flexible and inexpensive solution," said Windham. "I'm honored to be delivering the keynote at ITEXPO East 2010 and sharing some of the great accomplishments of open source in the enterprise with attendees."

Registration for ITEXPO and Digium|Asterisk World remain open. Vendors interested in participating in ITEXPO or Digium|Asterisk World should contact Joe Fabiano at 203-852-6800 x132.

In a fascinating deal, IBM and Digium announced today that they are teaming up to offer Asterisk for Smart Cube, a customized version of Asterisk Business Edition. IBM's Smart Cube is very similar to Microsoft Small Business Server (SBS), a pre-packaged bundle of various IT & business applications - except in this case Smart Cube is based on Linux not Windows.

Businesses using Smart Cube can be up and running with a complete IT solution to run their business, including the operating system, integrated middleware, database, security and back-office functionality such as file, print, backup and recovery, and more. Extending the IBM Smart Cube to IP telephony and unified communications is a natural extension of the Smart Cube.

Asterisk for Smart Cube has administration capabilities built right into the Smart Cube Smart Desk GUI, which is perfect for the SMB. Asterisk initially ran on rPath but now uses the very popular CentOS distribution. IBM on the other hand is very partial to SuSe Linux. thus one of the technical challenges IBM and Digium worked on was getting Asterisk Business Edition to run on the SuSE Linux platform. Additionally they worked on seamlessly integrating it into the Smart Desk GUI.

How this affects Digium's own home-grown Switchvox SMB offering remains to be seen. But Digium gaining access to IBM's huge distribution and reseller channel is great news for Digium.

Via internetnews.com

The 10th Anniversary of the AstriCon 2009 conference is next week in Arizona (October 13-15) and it is shaping up to be a great show. For one, AstriCon is sporting 30% more companies in the Expo Hall than last year's conference. The AstriCon organizers have even thrown in some cool freebies to add to your Asterisk arsenal.  All attendees receive three popular Asterisk licenses, including: Skype for Asterisk, Fax for Asterisk, and G.729 for Asterisk.

AstriCon 2009 will be held at the Renaissance Glendale Resort and Spa near Phoenix, Arizona. You can register for the conference at www.astricon.net.

Check out the quick video recording I made of two Polycom VVX 1500 IP video phones making a test video call. I used an iPhone 3GS to capture the video of the test call between the two phones. The iPhone's video recording quality isn't too shabby, but doesn't truly give you an idea of the quality of the Polycom's VVX 1500 touchscreen. (Note: The video phone was tilted far back, so it was aimed directly at the fluorescent lighting in the ceiling. So you see some whiteout when the VVX 1500 camera gets blinded by the ceiling lighting.)

Having a touchscreen IP videophone is pretty cool and I'm enjoying testing it so far. This is just a quickie video demo. I hope to do a full review of these phones very soon. Till then, enjoy the Youtube video below:

After ITEXPO's resounding success in Los Angeles (over 6,000 attendees), we can definitively say VoIP hasn't been as badly affected as other industry sectors within the U.S. economy. In just about 3 weeks, we can confirm this is true with the big AstriCon event held at the Renaissance Glendale Hotel & Spa in Glendale, Arizona. TMC's Internet Telephony Magazine is a media sponsor for the event. Companies participating include Aastra, Adhersion, Digium, PIKA, Polycom, Sangoma, Xorcom, and more

This marks AstriCon's sixth year as the official conference for Asterisk, the world's leading open source PBX. According to the event organizers, "AstriCon's mission is to expand awareness and knowledge of Asterisk over the course of a three-day conference and exhibition. AstriCon includes a wealth of information for every Asterisk user, whether you are getting started or have already discovered the power of Asterisk."

I for one am a little sad I won't be going. Too much stuff to do back at the home office. TMC is growing like gangbusters and we are very close to moving into a state-of-the-art facility. Now imagine you are in charge of moving TMC's entire data center to this new facility with minimal downtime. It's enough to keep any CTO awake at night. It's not happening for a couple of months, but this will require some massive planning by me and my team.

I tested Siphon, a SIP-based VoIP application for the iPhone, in California at ITEXPO. Interestingly, Siphon worked perfectly in California over AT&T's 3G data network. Yes, you read that right - VoIP over 3G! I couldn't contain my giddiness when I realized I could now register my iPhone with TMC's Asterisk-based IP-PBX and make/receive calls. I've tried Siphon a few times in the past and it never worked over 3G - only WiFi. I thought perhaps AT&T was now easy their restrictions and allowing it. (silly me)

However, once back in Connecticut I tested it again and it didn't work. Apparently, in some parts of the country AT&T is blocking port 5060, the default SIP port. I did some port testing on my iPhone and indeed AT&T is blocking outbound port 5060. While I was in Los Angeles I was able to use Siphone to make & receive VoIP calls over the 3G data connection through my corporate Asterisk-based PBX. I was able to receive calls to my TMC extension as well. Guess it was good while it lasted...

Now, Siphon does let you change the local SIP port from the default 5060. In theory, the Siphon application can be modified to use a different outbound port and then you could setup some port forwarding rules on your firewall, i.e. map the 'always open' port 80 (web) on your firewall to forward to port 5060 when connecting to your SIP-based IP-PBX's IP address. Or if you IP-PBX is already using port 80, there are plenty of other outbound ports that AT&T doesn't block.

Apple has rejected and blocked Siphon from the App Store. Interestingly, Apple allows other SIP clients (WiFi-only) to be downloaded from the App Store, including iPico, fring, iSip (supports push notifications of calls), Acrobits Softphone, WeePhone SIP, and Nimbuzz. What's interesting about the Siphon app is the whole saga the developer had to go through with Apple when submitting this SIP application to the App Store. It wasn't pretty...

The short story is that even when Siphon didn't support VoIP over 3G a few versions ago, Apple still rejected the app providing a lame excuse. Then after several attempts, Siphon went "underground" and provided their SIP app to Cydia, the primary jailbroken app store - with full VoIP over 3G functionality. If you can't beat em', screw em'! That's why a lot of apps have gone to the Cydia App Store to get around Apple's ridiculous restrictions.

Check out this screenshot of my iPhone showing how you can enable Siphon over EDGE/3G:



Unfortunately, like I said earlier, AT&T is blocking outbound port 5060 in some parts of the country, so simply enabling Siphon over Edge/3G by itself won't work if they block it. Apparently, the AT&T cellular network in Los Angeles, California works though. If anyone else has gotten Siphon to work over the AT&T 3G network, post a comment - or even if it didn't work. Would be a good gauge of how widespread they allow/disallow this.

The day is coming when the carriers will have to allow VoIP over 3G. Look at what VoIP, and especially Vonage did to the traditional landline industry. We went from paying long distance minutes by the minute to an UNLIMITED plan with UNLIMITED minutes for a flat rate. The mobile industry will soon have to follow suit.

In fact, the first wireless carrier that lets me register my cell phone to my SIP-based IP-PBX over a 3G data connection will become my new wireless service provider and have my business. I'm sure millions of others feel the same. Heck, charge me a few cents for terminating or originating my SIP-based calls. I'd pay for the ability to use my corporate identity (CallerID) when making business calls on my personal cell phone. Or just count SIP calls as 1.5x or 2x per minute of usage towards my current monthly plan's bucket of minutes. Of course, the carriers would have to detect when a SIP call originates or terminates, which is a technical challenge. They'd have to do packet inspection on a mass scale to support this.

Still, there has to be an appropriate revenue-generating business model for the wireless carriers that will allow their customers to use SIP over 3G. Make it $5/month extra or something. Vonage took the traditional landline providers by surprise, causing the defection of millions of users. So if the wireless carriers wait too long, some new wireless carrier is going to come along and do the same by offering VoIP/SIP over 3G. You mark my words...

At ITEXPO, Aastra announced G.722 wideband audio codec support (HD audio) in their new 67xxi firmware version 2.5.0 or later. It's available as a free download from Aastra with no strings attached. No need to upgrade your 67xxi (formerly 57XXi) phone to a newer model to get HD audio. I spoke with Aastra at ITEXPO about this free upgrade and got a demo as well. First, it's important to note that the speaker and the microphone built into the existing 67xxi don't have the full frequency response for full HD audio. However, there is still a noticeable improvement in audio quality, especially in the low-end bass side. For 100% HD audio from the microphone to the speaker to the full-range frequency response you can purchase their newer phones which feature upgraded hardware components.

But for the thousands of Aastra phones out there - including one I use as my primary desk phone - you can simply upgrade to the new firmware and immediately see a performance improvement. I also mentioned to Aastra some issues with the speakerphone switching to half-duplex mode when there is a lot of ambient noise or if the remote caller is talking too loud. The remote caller who is speaking can't hear you (half-duplex) no matter how loud you yell. Well, apparently this new firmware does some tweaks and solves that issue as well.

Aastra is calling their wideband audio feature Hi-Q. Let's face it, Polycom has done a good job positioning themselves as having the best sounding IP phones, especially with their HD Voice product line. Aastra's new Hi-Q offering now allows them to compete with other HD phones including not only Polycom, but also Cisco, Snom and others. Aastra Hi-Q wideband audio will be supported on the 6757i CT, 6757i, 6755i, 6753i, 6751i, 6731i and 6730i.

I did a demo with Aastra on the ITEXPO show floor with Hi-Q turned off and then on and I noticed the difference right away. Can't wait to get back to Connecticut and upgrade my Aastra 6757i CT phone! They also demo'ed some cool new DECT 6.0 phones and new WiFi phones that are just now coming to the United States. (They are currently available in Europe)

According to Aastra, "Aastra's Hi-Q audio technology is a software based acoustic optimization, backwards compatible with existing 67xxi series SIP phones, delivering a more life-like conversation and richer user experience via an industry standard G.722 wideband codec."

Aastra 67xxi firmware with Hi-Q wideband audio support can be downloaded here.

At TMC's ITEXPO, Digium and Skype announced the official launch of Skype for Asterisk, which was launched as a closed beta back in September 2008. Well, now anyone can now download Skype for Asterisk and make & receive low-cost calls leveraging Skype.

According to Digium, "Now businesses can take advantage of Skype's low-cost calling to landlines and mobile phones and free calling to more than 400 million registered Skype users around the world. Skype for Asterisk allows businesses to access the world's largest community of people communicating over the Internet, natively encrypts all voice calls and lets companies manage their Skype user accounts via Skype's Web-based Business Control Panel. Businesses already using an Asterisk-based phone system can add Skype as another complementary form of communications by downloading Skype for Asterisk, without additional costly hardware. Skype users can benefit from the advanced call features of Asterisk, including call transfer, interactive voice response, automated call distribution, flexible call-routing and many more."

"Digium has been using Skype for Asterisk for the past few months while the product has been in development," said Danny Windham, CEO of Digium. "We created Skype accounts such as Digium Sales and Digium Support--a convention I suspect many companies will quickly adopt. Now, our customers all over the world can call us for free using Skype and our Asterisk PBX processes the inbound call just like it would a normal call. This is going to save Digium and our customers a lot of money."

DATUS Corporation is a Digium Select Partner with nearly four decades of experience designing and implementing communications networks in Germany. The company has nearly completed an Asterisk installation at 2,100 sites for LVM Versicherungen, a major insurance firm, and also works with Digium to design features for Asterisk that are of particular interest to European businesses. "Adding Skype for Asterisk to the DATUS indali OBX, our IP-PBX, will offer our customers inexpensive and secure international calling that, for instance, could be used for toll free customer services," said Jonny Kueppers, vice president of sales and marketing at DATUS. "We believe that the price and cost savings will be welcome with today's budgets."

"The combination of Skype and Asterisk gives those companies that have relied on Skype the advanced call management capabilities of Asterisk, while Asterisk users get free calling to more than 400 million registered Skype users and low Skype rates when calling landlines and mobiles," said Stefan Öberg, vice president and general manager of Skype for Business. "We believe the product will bring together two of the largest groups of users that value flexibility and cost savings in their PBX systems."

Foehn Ltd, Digium's U.K. Solutions Partner, has been designing and implementing Asterisk-based solutions for more than five years. The company's technical director, James Passingham, commented: "With Skype for Asterisk, we can offer our clients even more freedom in business communications. The ability to unlock the lower call costs of Skype provides a huge savings opportunity, especially for those with offices and customers around the globe."

Skype for Asterisk Features
Skype for Asterisk, which is compatible with the free and open source Asterisk versions 1.4, 1.6 and AsteriskNOW™, as well as the commercially licensed Asterisk Business Edition™, is unique in the market today. It is the only solution that integrates directly with Skype, enables multiple concurrent Skype calls from a single Skype account, and supports both G.711 and G.729a calling.

• Make Skype-to-Skype calls.
• Receive calls with online numbers (SkypeIn).
• Make world-wide PSTN calls to landline and mobile phones (SkypeOut).
• Make and receive multiple concurrent Skype calls from the same Skype account.
• DTMF support for incoming and outgoing calls.
• Read Skype profile fields from incoming calls.
• Set and retrieve online status.
• Set privacy settings.
• Handle incoming Skype calls using Asterisk applications such as voicemail, ACD, MeetMe conferencing, etc.
• Simultaneous access from both Asterisk and the Skype desktop client.
• Trunk calls between Asterisk servers over Skype.
• Supports G.711 and G.729 (included) codecs.

SIP Print is announcing today the general availability of a new, enterprise-class call recording platform for mid-market enterprises. The new SIP Print SME platform offers support for up to 200 seats per location, along with RAID hot-swappable drive bays, dual hot-swappable power supplies, and a Core 2 Quad Series processor.  Today's announcement is being issued in conjunction with TMC's ITEXPO Conference in Los Angeles.
 
According to SIP Print, SIP Print SME is a new, more powerful appliance designed for the needs of small and mid-size enterprises, or any organization with the requirement to record up to 200 seats per location.
 
"We introduced our highly affordable SMB product one year ago to meet the needs of small business with a need to record calls for training, QA, or compliance purposes, but simply couldn't justify the expense or hassle of the legacy recording systems on the market," said Jonathan Fuld, CTO for SIP Print.  "Since that time we've seen tremendous demand for a similar, but more powerful system in the mid-market enterprise arena.  We're pleased to introduce SIP Print SME as the ideal solution for mid-sized enterprises with the need for a system that is easy to install, easy to use and maintain, and easy to afford."
 
SIP Print SME is a 1U appliance and is certified as compatible and interoperable with many of today's leading IP PBX systems, including: Allworx, Aastralink, ADTRAN, Altigen, Avaya Distributed Office, Cisco, Epygi, Fonality, Grandstream, Mitel, NEC 8100, NEC 8300, Nortel, ShoreTel, SIPfoundry, Toshiba, Zultys, 3Com, and more. As configured, SIP Print SME is capable of recording and storing the equivalent of one handset, 24x7 for 15 years.

Check out my recent review (last month) of their previous SIP Print appliance which I gave extremely high marks.

Can you believe ITEXPO is just two weeks away? It's also almost September. Where did the Summer go?

ITEXPO, the #1 VoIP conference in the U.S., has several educational tracks you might be interested in checking out. Of particular interest to me are the two separate Asterisk and the Switchvox training courses. As Asterisk's popularity continues to grow, so does its development and complexity. Last year's Asterisk isn't the same as this year's, so it's never too late for a refresher or to learn about the newest features.

I've only seen demos of Switchvox and haven't actually put Switchvox through the full test-drive ringer, so I might want to check out the Switchvox training course just to see what's new and what's different from regular Asterisk. Also can't hurt to learn how to use and manage it since I'd like to review it at some point.

You can check out and register for one or both courses here.