January 2008 Archives

Who hasn’t been on an important conference call and heard the infamous “buzz” sound on the speakerphone caused by cell phone interference. Nextel two-way “push to talk” phones are notorious for causing this “buzz” sound, as well as Blackberries. Still don't know what I'm talking about? Sure you do. Just play this MP3 file and you'll recognize this annoying buzz sound.

As I commented to Polycom, there are occasions when I'm watching FOX News or some other live TV program and I hear the "buzz" sound during the TV show that I know is coming from their end. How unprofessional and very annoying. You'd think professional news studios would have shielded microphones, or the news anchors would turn off their cellphone while on the air, but I digress.

In November of 2007, I reviewed the Microsoft Response Point IP-PBX. One of the Response Point's main features is the "Easy button" that you can press and then speak commands to the RP system, such as "transfer to Tom Keating" or "call Bill Gates".

Well yesterday, I received a news tip about an Asterisk solution that duplicates this "Easy button" speech-rec functionality. The freepbx.org blog explains:

... I saw a video on the Response Point "magic button." A voice recognition button for a PBX? It appeared that Microsoft really did something here.

I wondered to myself if Asterisk could do it. A while back I utilized the LumenVox speech recognition software for Linux/Asterisk to build a speech enabled company directory for Asterisk. That was pretty cool, but a magic button would be a Killer App.

So Ethan Schroeder, an Asterisk fan, set on a course to make a magic/easy button for FreePBX, an Asterisk-based solution.

He writes:

From the start of a dream to actual implementation was quite an experience. Creating a button that works when you are not in a call for some functionality (call initiation) and works while you are IN a call for call control turned out to be a difficult task, but I managed to make it work.

The result is a magic button that when pushed plays a fun tone and let's me speak to my phone system in wondrous ways:

• "call John Smith" or "dial John Smith" - dials by name (John Smith, John) or extension number (Four-thousand-one/4-zero-zero-1/4'oo'1), or even speak the digits of a 7, 10, or 11 digit phone number.
• "Transfer to John Smith" - transfers a call to a name or extension.
• "Transfer to John Smith's Voicemail" or "Transfer voicemail John Smith" - transfers the call directly into John Smith's voicemail.
• "Park call" or "Park caller", parks the call and announces the parking slot.
• "Retrieve calls" - queries asterisk for all the parked calls and gives the user their options using the Flite text-to-speech engine (which I'll soon be switching over to the incredibly cool Cepstral engine with their new "Asterisk Allison" voice).
• "Retrieve call [parking slot]"- Retrieves a specific parked call.
• Needless to say, that’s only the beginning!

So you're probably wondering how you get your hands on this great application? Well, I have good news for ya. Ethan is giving it away to all participants of the FreePBX training taking place in South Carolina in a few weeks, as seen here:

Want to see this in action, and even take home the technology? I'm demonstrating it all to participants of the FreePBX training when I present in South Carolina February 27-29th, 2008. I'll also be giving away all the parts to make the magic button work, the AGI, LumenVox configs, Asterisk dial plans and an Asterisk 1.2.x patch required for the parking feature, back ported from 1.4. All you need is a LumenVox starter kit and it looks like you’ll get that too.

I didn't know FreePBX was offering training courses, so this is news to me. I wish them well and hope it is as successful as the FtOCC (Fonality trixbox Open Communication Certification) Training at ITEXPO that was just held last week. 52 people were in the trixbox training class -- all with laptops learning how to setup & configure the Asterisk-based trixbox IP-PBX. I should ping FreePBX and see if they want to co-locate their training at ITEXPO since we draw a lot of Asterisk enthusiasts to our show. AFAIK, it doesn't cost them anything either, so it's a win-win.

8x8's Packet8 may not be the largest VoIP service provider, but they have nice stable growth. Unlike some VoIP service providers which tried to grow fast but then crashed and burned (See SunRocket posts here, here, and here), Packet8 has had slow steady growth over the years.

So I'm pleased to report that 8x8, Inc. (Nasdaq: EGHT), provider of Packet8 broadband Voice over Internet Protocol (VoIP), videophone and mobile VoIP communication services, today announced revenues for the third quarter of fiscal 2008 were $15.8 million, compared with $14.8 million for the second quarter of fiscal 2008 and $13.3 million for the same period of the prior year, an increase of 7% over the prior quarter and 19% year over year.

Net income for the quarter was $1.4 million or $0.02 per share, the second highest quarterly earnings in the company's history, compared with a net loss of $2.6 million or ($0.04) per share for the second quarter of fiscal 2008 and a net loss of $2.8 million or ($0.05) per share for the same period last year. Total revenues for the nine-month periods ended December 31, 2007 and 2006, were $45.3 million and $38.7 million, respectively, an increase of 17%.

You can read the full financial results after the jump...

Recently I wrote about the challenges of Fax over IP and T.38 real-time faxing and the lack of good T.38 support in Asterisk. Though I did point out a few fax workarounds in Asterisk. I received a flurry of emails and comments. One stated:

There actually is full fledged t.38 support for asterisk: http://www.attractel.com/faxterisk.php

2 week trial licenses are available.

Also have a look at our zoiper softphone which can receive, and in the next version, somewhere next week also transmit faxes over t.38.

I checked out the Faxterisk commercial software solution and it seems like a good T.38 solution for Asterisk. Here's how it works: a fax terminal passes analog data to a T.38 gateway. The gateway takes the signal, demodulates it and transmits it over the existing IP network to another T.38 capable device(e.g. Internet Aware Fax(IAF) or a compatible Analog Telephone Adapter (ATA) or another T.38 gateway that modulates back the digital signal and sends it over the PSTN to another fax terminal.

Faxterisk bold claims, "The sensational part of our Faxterisk solution is that it is actually a far more reliable faxing method than traditional analogue faxing. This is not a phenomenon; it is due to the fact, we may want to believe that landlines are good carriers, but this is far from true as far as greater distances are concerned. IP networks are more reliable carriers over long distances and with our FoIP solution, you will inevitably enjoy faxing more than ever!"

The Faxterisk package includes:
T30faxRX for receiving normal faxes over PSTN
T30faxTX for transmitting normal faxes over PSTN
T38faxTX for T.38 fax origination (IAF fax)
T38faxRX for T.38 fax termination (IAF fax)
T38gate for FoIP gatewaying

You don't even have to change anything in your dialplan. Basically, Faxterisk auto-detects the fax signal and automatically switches to fax mode. I should mention that Faxterisk works with Asterisk 1.4.X versions only. It supports the following ITU-T modems: V.27ter, V.29 and V.17 with speeds of up to 14400 bps.

Alternatively, Spandsp adds integrated faxing capability to Asterisk via some GPL patches. Spandsp is a library for Digital Signal Processing(DSP) which is used to turn images into fax tone audio signals and back. You can look at this instruction link and the download link for some help installing. Spandsp only supports V.29/9600bps due since higher speed modems for fax are encumbered by patents. spandsp implements only unencumbered standards, so it may be distributed freely.

Last week TMC held our very successful ITEXPO tradeshow covering various IP communications and contact center technologies. One thing we all need to keep in mind when it comes to voice over IP is that voice over IP is simply a dumb transport. What really gets customers excited and coming to various VoIP tradeshows is to learn about how they can leverage VoIP within their business processes, whether that's screenpop of customer data, remote agents using soft or hard phones, mobile phone capabilities, etc. Customers are looking to integrate all of their various communications - email, voice, CRM, etc. VoIP is definitely becoming less about saving on toll charges and more about the applications.

What's nice about this last ITEXPO was that it had a nice mix of IP communications companies and call center companies. Call centers are often at the forefront of communications technologies since they need to optimize their communications and make their agents as productive as possible. TMC's Customer Interaction Solutions Magazine is the premiere magazine discussing call center technologies and how VoIP is changing the game in the call center arena. So it certainly didn't hurt that CIS Magazine (through active promotion) helped bring call center companies and call center-related attendees to ITEXPO.

In any event, let me tell you what I found interesting at the show. One interesting company, and whom I know quite well, is Intuitive Voice Technology. I stopped by their booth at ITEXPO to check out their Asterisk-based Evolution PBX 3.0 platform. I like the Evolution PBX system a lot. It has some unique features that you won't find in any other Asterisk solution. At ITEXPO they were demoing the integration of Evolution PBX with a Citel Portico TVA gateway, which enables you to connect legacy digital phone handsets. Useful for instance if the receptionist has a legacy phone with lots of programmed extensions for quick transfers. Why buy a new expensive VoIP attendant console when the old one works just fine? You do pay about $139/port on the Citel gateways, but $139 is still cheaper than most VoIP phones, so there is some cost savings there.

Another cool feature in the Evolution PBX is that it supports call screening. Callers can be prompted for their name and then you can accept the call, send to voicemail, and I think they mentioned you can redirect to another number. I use call screening all the time since I get public relations pitches all day long. I wouldn't get anything done if I actually took every phone call. I inquired about the ability to screen a voice message being left and then 'pull' the caller out of voicemail if you deem it necessary. Similar to what you can do with most home answering machines. Evolution PBX doesn't currently have that capability. We discussed how the voicemail system in Asterisk v1.2 wasn't really designed for this level of customizing to add additional functionality. However, with Asterisk v1.4 the voicemail system has been revamped it is much easier to "hook" into the voicemail and develop powerful applications such as voicemail screening and pulling callers out of voicemail.

XORCOM is another interesting Asterisk-based solution called Astribank. I've nicknamed them the Analog Asterisk due to their main "claim to fame" of supporting the most analog trunks in a single Asterisk appliance. They support FXO & FXS ports, but they also support BRI (ISDN) and PRI as well. They actually leverage the USB channel to daisy-chain multiple appliances to add additional scalability. Neat idea! Astribank comes in two flavors - a compact chassis and a standard 1U 19" rackmount or wall mount chassis.

Now let me get off my Asterisk kick for a moment and tell you about MobileMax, one of the coolest mobile applications you've never heard of. I kept trying to stop by their booth, but every time I did their was several people talking to just a single guy manning the booth. It wasn't until the very final hour of the show that I was able to stop and check out MobileMax. First, let me point out that one of the main challenges when using Jajah, Rebtel, Talkster, and all the other mobile apps that aim to reduce costs is "usability". In some cases, you have to install an application, while in others you have to go to a web browser. In either case, you are forcing users to use another interface to initiate dialing. Why can't you simply dial a phone number and let the phone figure out the best way to dial and route the number?

Well, with MobileMax, you can do just that. Using MobileMax’s technology, mobile subscribers can dial international calls directly from their mobile handset — with no additional keystrokes and automatically enjoy reduced-rate services. Their solution allows end users to dial with their cell phones as they are used to through their address book or call log. Their application will then route the call to a specific access number on a PBX or IVR. The MobileMax client resides on the cell phone and will intercept your dial string based on pre-defined criteria such as destination number, caller location (in/out home country/home area code), connectivity (WiFi or 3G availability & quality), VoIP provider plan (QoS, termination cost), and more.

Perry Nalevka, Director of Business Development gave me a demo of the application in action. From his personal cell phone he initiated an outbound phone call like he would normally do. After pressing the Talk button, the phone call is initiated (said 'calling'), but a second later, the phone call was stopped in its tracks by the MobileMax application and then the call was placed through the MobileMax application. Service providers, pre-paid carriers, etc. using MobileMax can divert the call to their platform's access number, send the PIN tones and the required destination number. All of this is seamless to the mobile phone user. Currently, MobileMax supports over 300 mobile phone models.

Another interesting company I saw at ITEXPO was iKNOWWARE, a web-based on demand process management/CRM solution that can be accesses from PCs, PDAs, mobile, and VoIP telephony products. Just like your typical SaaS hosted CRM platform, iKNOWWARE includes contact management and case management, but they also integrate inventory control, order entry, and project management. They showed me how well their application works even on small mobile phone screens.

GVSC USA was demoing some cool VoIP/videophones in their booth, including the Leadtek BVP8882 videophone and the Leadtek NetWork Care Phone NCP3680 surveillance camera. They demonstrated how the BVP8882 could not only be used as a videophone, but it can be used to watch TV channels over the Internet and even remotely change the channel. Benny Shea, the President showed me how he was able to change the channel on a TV tuner located in South Korea. The quality was pretty good and it supports H.261 and H.262. The Leadtek NetWork Care Phone NCP3680 is a very thin camerphone designed to stick outside your door. It has a built-in speakerphone and two way audio plus one-way video, along with a one-touch button to initiate audio/video communication with a PC-based softphone application.

Here are the phones if curious.

Leadtek BVP8882 videophone

Leadtek NetWork Care Phone NCP3680

All in all, it was a great show and everyone was ecstatic with the attendance. I saw some other cool stuff like the new Adtran 706 and the Adtran 712, which marks Adtran's entry into the VoIP phone biz. The phones are SIP-based, feature PoE (Power over Ethernet), dual RJ45 network ports, and more. SmithOnVoIP took a photo (below) of the phones at ITEXPO and has some insights on these new phones worth checking out.

I came home last night and learned my 21-month old daughter Megan had decided to take our Canon Powershot G5 digital camera off the kitchen table and plop it into the dogs' water bowl. My wife scrambled to remove it quickly, but the viewfinder is now all fogged up and it won't turn on. Maybe it will dry out, who knows?

I joked with my wife that surely Megan didn't get her "gadget destroying genes" from a gadget lover like me. Then again, my background is in engineering and engineers like to see how things work, and they love to stress test something to the max, etc. So perhaps the little engineer in her was simply stress testing my digital camera. Well, Megan, it failed - it failed miserably.

Ah well - it was time for a digital camera upgrade anyway.

On January 14th, I joined my local gym's The Biggest Loser competition. I promised periodic updates, so here goes.

Week 1 (Jan 14th) - weighed in at 198lbs
Week 2 (Jan 21st) - weighed in at 189lbs (-9 lbs) 9lbs lost in 1 week - Woohoo!
Week 3 (Jan 28th) - weighted in at 189lbs (0 change) - I was extremely sick 5 days before and was attending ITEXPO, so didn't get a chance to workout. I'm expecting at least a 4lb loss this week. Wish me luck!

Mig33 is a mobile instant messaging application running on Java, which means it runs on over 250 mobile phones. It has mobile VoIP service, chat, chatrooms, SMS and instant messaging. Mig33 also works with MSN, Yahoo!, ICQ , AOL, and Google Talk. They claim over 9 million users even with their non-descriptive company name. Sounds like a Russian fighter jet to me.

Anyway, today Mig33 announced closing a Series B funding of $13.5 led by DCM with participation from existing Series A firms Accel, Redpoint and TVP. mig33 will use this round of funding to continue expansion of the company and growth of the mig33 global communication service in the U.S. and around the globe.

Since receiving Series A funding, mig33 relocated the entire company to the U.S. They claim their community has more than 2 million sessions per day, sends more than 45 million messages each day, and shares more than a million pictures a month. mig33 offers a mix of free and inexpensive services, including VoIP calls, instant messaging, e-mail, text messaging, picture sharing, social networking and community features.

To access it simply go to wap.mig33.com on your Java-enabled phone.

According to NetworkWorld, "Digium, the business founded by Asterisk creator Mark Spencer to capitalize on his PBX, says it plans later this year to release a new version that will support much larger deployments. New Jersey-based service provider VoicePulse plans a hosted PBX service based on Asterisk deployed on virtual servers."

Reminds me of my 2006 VoIP Predictions post where I said:

Asterisk (Digium), the open-source IP-PBX gets deployed by a Fortune 500 company. I should point out that I don't mean a small Fortune 500 branch office that installs a single Asterisk server. I'm talking about a Fortune 500 company that installs Asterisk in at least 5 branch offices OR at their main corporate headquarters. This would be major news for Asterisk and another endorsement of the open source movement. Cisco is asked for comment on this young upstart and simply replies, "Cisco is a well established company with reliable, scalable, and affordable VoIP solutions that is committed to its customers and will be here for years to come."

I'm still waiting for that first Fortune 500 deployment. It's now been two years since I made the prediction. C'mon Asterisk, I've been such a faithful follower of all things Asterisk, don't let me down! You can do it!

T.38 is an ITU standard for sending FAX across IP networks in a real-time mode. T.38 faxing has been hyped a lot, while others pondered if T.38 was dead. Regardless of which camp you stand in, we can all agree that T.38 hasn't exactly taken off, including on the Asterisk IP-PBX. Faxing support in general on the Asterisk platform has been sorely lacking and one of Asterisk's weaknesses. Though there are some workarounds such as Hylafax and IAXmodem. IAXmodem is used to interface Asterisk with HylaFax.

However, as of Asterisk 1.4, Asterisk allows the passthrough of T.38 UDPTL data. Asterisk does not actually decode the T.38 protocol, it simply passes the T.38 data to a T.38 PSTN gateway. For termination you'd need something like the Gridborg HMP Server or CallWeaver. I should also mention that AudioCodes has helped write articles for TMC as well as held webinars on TMC discussing T.38 networking faxing used in conjunction with SIP, which are worth exploring. AudioCodes also makes voice TDM hardware which runs on the Asterisk platform.

The other issue with T.38 is that T.38 capabilities vary greatly between VoIP providers. Some do not support it at all, while others support it fully, up to and including V.34 FAX (high speed 33,600bps faxing). Similarly, some IP-FAX hardware supports some capabilities of T.38, but not all. Most don't support the higher speed V.34 spec.

One interesting solution that was launched last week at ITEXPO was a new V.34 T.38 Fax-over-IP endpoint and gateway from Dialogic. They claim to be the industry’s first Fax Over IP Endpoint and Gateway solution supporting V.34 T.38. In theory you could use the T.38 passthrough function in Asterisk v1.4+ to send the V.34 T.38 fax traffic to the Dialogic fax gateway. Still, I'd like to see an integrated solution on Asterisk for sending/receiving faxes. One less server to manage plus less electricity utilization. We want to be green don't we?

In any event, here's the Dialogic release in case you missed it:

Dialogic Corporation ("Dialogic"), a global provider of world-class products and technologies for media and signal processing, today announced that it has released the industry’s first Fax Over IP Endpoint and Gateway solution supporting V.34 T.38.

The award winning Dialogic® Brooktrout® SR140 Fax Software and Dialogic® 3000 & 4000 Media Gateway Series (DMG3000 & DMG4000) enables enterprises to deploy fax server solutions confidently into existing Voice over IP (VoIP) installations, while leveraging their existing IP infrastructure. The addition of V.34 T.38 allows these products to transmit fax data at 33.6 kbps, surpassing the rest of the industry which currently transmits fax traffic over T.38 at a maximum of 14.4 kbps.

“The addition of rapid V.34 speeds to our highly reliable Brooktrout SR140 T.38 Fax over IP based fax server solutions is a significant break through in our industry, said Tom Linhard, President at FaxCore, Inc. A FoIP endpoint to gateway solution supporting V.34 T.38 will allow FaxCore and Dialogic’s mutual customers to send and receive faxes much faster. This will save them money by lowering both network bandwidth and telecom expenses.”

Dialogic Corporation is the market segment leader in fax and fax-over-IP platforms, offering one of the fullest featured and broadest fax solution portfolios on the market today. The Brooktrout SR140 and DMG3000 & DMG4000 platforms support real-time FoIP, providing companies with the ability to integrate fax servers into their VoIP network. In addition to capturing the benefits of IP-based communications, these products have excellent performance, reliability and scalability, attributes that enterprises have come to expect from the Dialogic Brooktrout Circuit Switch based fax product line.

“Dialogic’s fax technology is the engine of the modern fax server that delivers document management, business process automation and compliance with industry mandates, such as Sarbanes Oxley and HIPAA," stated Jim Machi, Senior Vice President of Marketing for Dialogic. By implementing V.34 T.38 fax on their IP network, enterprise customers will not only benefit from the savings achieved by automating fax document processing and ensuring the integrity of internal control procedures and transactions for regulatory compliance, but can also expect to have lower operating costs than if they were using comparable FoIP solutions currently on the market.”

Another excellent ITEXPO show down here in Miami. All the exhibitors, attendees, and conferees seemed very happy. But the happiest guy has to be the convertible Ford Mustang GT raffle winner, as seen here:

Today, at Internet Telephony Conference, Iwatsu Voice Networks (Booth #617) announced the expansion of its Valued Integrated Partner program (VIP) with the addition of a new partner, Ingate Systems. Iwatsu's VIP program is designed to offer products and services that provide value-add to the Iwatsu Enterprise Communications Server (ECS), Iwatsu's flagship call processing solution for small- to medium-sized businesses.

Ingate will add its Ingate SIParator line of products to the Iwatsu VIP program. The Ingate SIParator is a device that connects to an existing network firewall to seamlessly enable the traversal of real-time SIP-based communications by solving the Network Address Translation (NAT) traversal issues with the protocol, while maintaining the security and integrity of the enterprise network. What's really nice about SIParator is that you can keep your existing firewall will all of your various firewall rules & policies, but you make the existing firewall "SIP-aware" by adding the SIParator. Personally I think they should call the product the SIPinator, but that's just the Terminator fan in me.

One of the unsung pioneers in the VoIP industry is Rich Tehrani. Rich started Internet Telephony Magazine in 1997, the first magazine to exclusively cover VoIP. Before Internet Telephony, Rich also launched CTI Magazine in 1994, a magazine focused on Computer Telephony Integration - the precursor to voice over IP. TMC Labs, which Rich co-founded with myself, did the very first VoIP review of the VocalTec software in 1996 - the first VoIP client.

Due to Rich's years of experience in telecom/telephony pre-VoIP and now in VoIP, he has met tons of CEOs and entrepreneurs in the VoIP industry. As such, Rich has a pretty good grasp of the industry and calls it the way he sees it, but it's rare for him to slam any company, so I was surprised to read Rich's rant against Sprint this morning. Rich strongly slams Sprint for the lawsuit that Sprint launched against Nuvox Communications, BroadVOX Holdings, Big River Telephone and Paetec Communications. Rich used to tell me "Always be friendly with companies even if they refuse to do business with you and they always spend money with the competition. Also be friendly with your competition. You should never burn your bridges with anyone. Things could change, I've seen it happen many times."

Guess that advice doesn't apply to Sprint. Here's an excerpt where Rich has a "call to arms" for the VoIP industry to fight against Sprint and other telcos that claim they invented VoIP:

Although I am not an attorney, it seems like it is time for the industry as a whole to come together and take on telcos who somehow believe they invented VoIP and a host of related technologies.

Anyone ever heard of VocalTec?

In a torrent of irony, these telcos are the same companies who tried to destroy VoIP for years before they started embracing it. What is the innovation -- bashing it until you start using it?

A couple days ago I wrote an article titled SIP Trunking is Hot! where I wrote about the SIP trunking session sponsored by Ingate, which was very heavily attended (75). A few hours later I noticed SmithonVoIP blogger Garrett Smith write a blog entry called No One Wants SIP Trunking. It was obvious to me he must have seen my blog post which was pro-SIP trunking and he was taking a counterpoint.

Before I even clicked through on my RSS reader to Garrett's article, I pondered why Garrett would take such a counterpoint. I know for a fact that there is a lot of interest in SIP trunking. A thought popped in my head that Garrett works for VoIP Supply which sells IP-PBXs and perhaps SIP trunking with hosted PBX functionality 'could' be a threat to VoIP Supply's business. Thus, perhaps Garrett was trying to downplay the importance of SIP trunking. But I know Garrett and I've never known him to have a hidden agenda in any of his writings. He speaks his mind and calls it the way he sees it, just like me.

So then I click through and read his article. The gist is that Garrett isn't anti-SIP trunking - just that he doesn't like the name or how it is marketing. Garrett explains:

Why do service providers and the industry at large insist on calling a phone line delivered over an IP network a “SIP Trunk”?

Customers do not want SIP trunks, they want business phone lines delivered via IP.

It is funny that no one outside of the industry knows what the heck a SIP trunk is, yet every company that provides IP based business phone lines continues to use this nomenclature.
...
...
Businesses do not know what a SIP trunk is and honestly they really do not care. They are not looking for SIP trunks.
What they are looking for is reliability, quality and low cost telephone service. The transport mechanism is secondary.

Garrett does have a point to a certain extent. Many customers don't care what SIP is and just want inexpensive calling rates over an IP connection. That said, there is huge interest in the VAR, reseller, and interconnect channel to learn about SIP trunking, the complexities, the QoS concerns & how to address them, who are the major players, SIP-aware hardware/firewalls, etc. My guess is the SIP Trunking Workshop sponsored by Ingate had a huge number of VARS/resellers/interconnects attending. I'm sure there were customers as well.

I ran into Dan York, formerly of Mitel and now with Voxeo told him about Garrett's post and I asked his opinion on SIP trunking. Dan was very high on SIP trunking and pointed out that the last ITEXPO's SIP Trunking session was also jammed packed. Update: Also just noticed Greg interviewed several VoIP players in his SIP Trunking Sessions Strikes a Chord post. Greg also seemed to notice like I did that SIP trunking is a very hot topic at ITEXPO. Greg writes, "I had the chance to speak with Olle Westerberg CEO, Ingate, Steve Johnson, president of Ingate, Anne Coulombe, who handles SIP product marketing for Avaya’s Solutions Marketing Division, Sean Rivers, channel account manager at Bandwidth.com, and Joel Maloff, the newly installed director of marketing at Bandtel." Go check out his blog for more details.

Garrett's blog title, "No One Wants SIP Trunking" isn't exactly true, since like I said there is huge interest in the channel. I do agree with him though that many customers don't care. That said, I'm glad TMC's educational conferences are at the very "top" -- training & educating the IP communications channel (VARS/resellers/interconnects) on SIP trunking. That knowledge will eventually "trickle down" to the customer when the channel explains to the customer the benefits of SIP trunking. Of course, the customer may simply filter out the technical details and focus on the bottom line. Maybe for the next ITEXPO we'll call it the "SIP/IP Trunking Workshop." Clear enough for ya Garrett?

While I'm away at ITEXPO covering all the latest VoIP news and meeting with industry insiders, my wife snapped some interesting photos of our yellow Labrador/Shepherd-mix dog Jessie with her paw on our 21 month old daughter, Megan. The look on Jessie's face says it all, but I'll translate for you just in case...


Jessie: Just stay down kid. It's a dangerous world out there and Dad's not home. Dad's away at ITEXPO learning about dangerous VoIP security exploits, and something about business SIP trunks. I do my "business" to tree trunks - not "sip" them, but that's another story. I know Dad is at ITEXPO covering other VoIP stuff that my dog brain just can't wrap itself around. Suffice to say, you're just too damn cute for your own good. I'm pretty cute myself and strangers touch me all the time. But don't worry, I got your back.


Jessie: [eyes darting towards stairs] Ut oh. Spider senses tingling... Oh wait, I'm a dog not a spider. Much be my super-sensitive ears hearing somebody pulling into our driveway. Stay tight kid. stay tight...


Jessie: I told you to stay tight! Don't get up kid! Those blue eyes are trouble.. trouble I say!


Megan: Thanks Jessie for watching over me while Dad is gone.
Jessie: It's a tough job, but a dog's gotta do what a dog's gotta do. Can I get a SIP of your grape juice?