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Skype for iPhone 1.1 Update
Late last night I saw an update notifications on my iPhone 3GS for Skype. New version? Sweet! I updated it and checked out the release notes for Skype 1.1 for the iPhone and iPod touch. It sports some new languages and now includes Danish, Dutch, Finnish, French, German, Italian, Japanese, Korean, Norwegian, Polish, Portuguese (Brazil), Portuguese (Portugal), Russian, Simplified Chinese, Spanish, Swedish and Traditional Chinese and of course English.
The bigest new feature is the ability to send text messages (SMS) using Skype credit. They also added voicemail support and improved dialing help when calling phones.
According to the official Skype blog, “we’ve made some improvements to the look and feel, particularly when calling phones using the dial pad.”
You can download the update in iTunes, or even better just launch the App Store on your iPod touch or iPhone and then tap Updates to get the latest version of all your apps!
Some things I would have liked to seen in this release:
- Push notifications (Skype might have to haggle with Apple on this one since Apple places restrictions on third party apps that can run in the background on the iPhone.)
- Video conferencing support - Though that probably isn’t coming any time soon.
So what would you like to see in a future release of Skype for iPhone?
Tags: apple, im, iphone, iphone 3gs, itunes, push notifications, skype, Skype 1.1 for the iPhone, voip
Related tags: skype iphone, calling phones, skype, iphone, release, update
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flaphone, the Flash Web-based SIP-to-SIP & SIP-to-Skype VoIP App, Adds New Features
Flaphone has done some updates to their Flash-based VoIP application. Back in 2007 I broke the story about the first Adobe Flash-based SIP VoIP app called Flashphone, later renamed Flaphone. Flaphone is a web-based SIP softphone that uses ubiquitous Flash (Mac, PC, Linux), to enable you to make or receive calls to/from all SIP phones and SIP services, including Yahoo! Messenger, MSN Messenger, and Google Talk. You can make free web-based Flash calls to Yahoo! Messenger, MSN Messenger, and Google Talk (gtalk) users. You can even make Flash-based SIP-to-Skype calls using Flaphone, which I tested back in February. You simply enter sip:skype_username@skype to make a call to a Skype username. Good stuff! 
They just announced some new features, including a new skin (white), emoticons in chat, and the ability to transfer files up to 5MB in size between Flaphone users if both have Flash player 10 installed. The file size is restricted for now, but Flaphone stated that when they add P2P support they will remove the restriction. I like Flaphone since I can run it from any PC and make SIP-to-SIP calls or SIP-to-Skype calls without installing anything. Definitely worth checking out.
Tags: adobe, chat, file transfer, flaphone, flash, google talk, im, msn messenger, p2p, sip, skype, voip
Related tags: flash based, messenger google, skype calls, skype username, yahoo messenger, flaphone
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sipgate enters U.S VoIP Broadband Market offering free calls
Today, sipgate is launching a new free VoIP broadband phone service called sipgate one. sipgate one is similar to Vonage, Packet8, and other broadband VoIP providers, but with some additional cool features and a fairly unique pricing plan. For instance, you get a free telephone number, no set-up costs and no monthly charges or minimums.
I spoke with sipgate CEO Thilo Salmon to find out more about sipgate one. First, unlike many VoIP broadband providers, sipgate one isn’t tied to any specific SIP hardware (locked ATAs, SIP phones, softphones, etc.), so you can use it with your favorite SIP device or use a SIP-based ATA and connect your favorite analog cordless phone. Want to use a softphone but don’t want to mess with SIP credentials? No problem - sipgate one has a free softphone app that will autoprovision for you. Those that want to use their favorite SIP device will be pleased to see a helpful drop-down list of many popular SIP devices with corresponding screenshots on how to configure the device to work with sipgate one.
What does it cost?
When using a VoIP phone, outbound calls to other sipgate users are completely free, just like Skype. However, inbound calls on the free U.S. phone number are also free, unlike Skype which I know charges a monthly or yearly SkypeIn subscription. I was a bit surprised sipgate was offering the first phone number for free, so I contacted Thilo a second time to confirm. He responded, “The first U.S. number is indeed completely free. So we are inviting everyone to sign up for a free number and as long as one only uses sipgate to receive calls on a SIP phone the service is free in its entirety. We do not even ask for a credit card. That does set us apart from Skype. While they do not charge per incoming minute, Skype charges a yearly (monthly?) fee as you have pointed out.”
When dialing other phone numbers in the U.S. and Canada are just 1.9¢ per minute and calls to toll-free numbers are free. Other rates apply when calling other countries. Thus, sipgate only charges for outgoing calls and E911 ($1.90 monthly), if activated. No other charges or fees are leveraged on a sipgate one subscriber. Thus, to get started you can initially charge your account with a minimum of $10 with automatic crediting if the balance falls below a user-specified amount ($5, $10, $20, $50). Thilo told me that they aim to keep their international rates lower than Skype’s to make them a very compelling option to potential customers. I then asked Thilo who was comparable feature-wise and he responded, “Google Voice. They don’t allow you to connect a SIP phone, but the features are very similar.”
The true power behind sipgate is its Web-based interface which gives you full access to your voicemail messages, recordings, and faxes. I tested a preview version of sipgate one and was pretty impressed with the features and ease of use. My experiences in my mini-review of sipgate follow below…
Thilo told me that they designed the Web-based interface with Google’s Gmail in mind. For instance, you can search, star/unstar a recording, as well as label recordings, which are very similar to Gmail. Additionally, the interface allows users to call someone back with a single click, as well as divert calls to other landline or mobile phones. Starting a call is as simple as clicking the New Call button and then choosing which phone device you wish to use. Below you’ll see I have 3 options for a new call - an arbitary connection (specify any phone number ad hoc), phone of Tom Keating, and Mobile or Landline of Tom Keating.
As seen by the screenshot above, you can bridge a call to any of your phone devices by initiating a call via the web. This is very similar to Jajah, who made the web-based calling method famous. This interface can be used on your web-enabled smartphone - Windows Mobile, Apple iPhone, etc. if the device doesn’t have an embedded SIP stack or 3rd party SIP softphone installed for making outbound calls.
Mobility features are just as powerful. You can have your home, office and mobile phone ring in parallel. If you are using a mobile phone with a SIP stack, (many Nokia series have SIP) you can be in the car, receive a VoIP call, and then simply press *6 on your mobile phone to record the call. When you next gain access to a web browser you can retrieve the call recording. This is perfect for sales people on the go and who need to enter important call details into a CRM system.
Other features included the ability to customize your outgoing message by uploading an mp3 file, divert calls to another number, create an ad-hoc conference with the touch of a button, and the ability to view missed calls with CallerID info. I asked Thilo what they use on the back-end and he told me they use a fork of SIP Express Router, which enables Class 5 type features. Additionally, Thilo stated there is no maximum storage limit for voicemail and faxes.
Outbound faxing is a breeze. You can click some fields on the web interface and add the fax number, from/to info, add a signature, and of course body text. You can also attach a PDF and it will render it. I asked Thilo about Word support and he said that was in the works.
Inbound faxing does require a separate phone number, but there is no fee to receive faxes. There is a nominal $2.90 one-time fee per additional phone number. I asked Thilo about CNG autodetection of fax tones using just a single universal phone number, but he said “We have found that most people don’t really like that.” Personally, I’d rather just have one phone number on my business card, but it’s a minor complaint.
“There is simply no barrier to people disconnecting their old phone lines anymore. Phone and cable companies have long been pushing voice plans in the region of $25 to $40 per month–which end up being as much as $60 or more with extra charges–and that’s just ridiculous,” said Thilo Salmon, CEO of sipgate. “Even with calls to other landlines and mobile phones, most users will spend less than $5 a month using sipgate one. And for those people only receiving incoming calls on their VoIP phone, the service is completely free.”
sipgate is also readying a multi-user edition of the service aimed at small businesses, which will not only replace landlines, but also customer premise phone systems. sipgate is certainly setting a new low-price benchmark with bundled powerful features that should cause VoIP fans to seriously consider them. Free phone number, free toll-free calling, free inbound calling, free inbound faxes, what’s not to love? 
Tags: broadband voip, e911, free calls, iphone, packet8, sipgate, sipgate one, skype, voip, vonage, windows mobile
Related tags: phone number, asked thilo, phone service, mobile phone, based interface, phone
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Incoming! for Skype on the Apple iPhone
A new Skype for the Apple iPhone app was just approved and is available in the iTunes app store called Incoming! for Skype. Basically, it makes every call an incoming call so you can save your minutes on your wireless dialplan, since they often feature unlimited inbound minutes and only charge for outbound minute usage.
Here’s how it works:
1. Open the Incoming! App on your iPhone
2. Dial the phone number, choose from Favorites, choose from address book or conference call
3. Press call and it will re-route the call through the Skype helper app on your home computer.
4. Your phone then rings, you pick it up and then the other person is called.
Using this app you can connect to any landline or mobile phone over Edge, 3G, and WiFi - unlike the ‘official’ Skype for iPhone app, which is WiFi restricted - unless you jailbreak your iPhone for 3G support of course! 
Basically the calls are routed through your home PC’s Skype software. You will use SkypeOut credits for PSTN calls. But if you’ve already signed up for one of Skype’s unlimited call plans (U.S./Canada $2.95/month), the call is essentially free. Another benefit is that you get some of the Skype features in this application. For instance, you can do 9-way conference calling on your iPhone, making business meetings while on the go a snap! Essentially, the app is harnessing the power of your PC for the audio mixing. (Note: I don’t believe the free ‘official’ Skype for iPhone app can do 9-way conferencing)
Features include a visual favorites list, address book and support for both Windows and the Mac. An iPod touch will work as well, but you can’t route calls to it, but you can route it to a nearby phone.
The app will cost you $4.99 on the iTunes store, which is pricier than most iPhone apps.
You can check it out here
Tags: apple, gadget, Incoming! for Skype, iphone, skypeout
Related tags: official skype, apple iphone, skype apple, skype iphone, incoming skype, skype
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Microsoft UC Developer Sandbox Featuring OCS 2007 R2 & Exchange Server 2010 Launches
Want your own little UC sandbox where you can learn how to develop speech and UC applications using Unified Communications Managed API (UCMA) without the hassle of setting up your own OCS 2007 R2 & Exchange Server 2010 beta environment?
Well, today Marshall Harrison over at GotUC.net announced a Microsoft UC sandbox for developers to play in, enabling them to develop UC applications quickly and easily.
Well, whatcha waitin’ for? Head on over and kick some sand in your very own UC sandbox!
Tags: beta, developer, exchange server 2010, Marshall Harrison, microsoft, OCS 2007 R2, sandbox, UC, unified communications, voip
Related tags: exchange server, sandbox
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50% of Mobile Voice Traffic Will Use End-to-End VoIP by 2019
Gartner, a respected research firm located just a stone’s throw from TMC said today, “mobile VoIP poses a huge challenge for traditional mobile voice providers.” You don’t say? Hmm, I would have never guessed such a thing. Ok, other than stating the obvious, the research does have some fascinating points, including claiming that ore than 50% of mobile voice traffic will be carried using end-to-end VoIP by 2019 - or basically 10 years from now.
“Mobile portal voice over Internet Protocol (VoIP) offered by third-party application-based providers poses a huge and direct challenge to the $692.6 billion global mobile voice market”, according to Gartner, Inc. Gartner predicts that over time traditional network-based mobile carriers face the real prospect of losing a major slice of their voice traffic and revenue to new non-infrastructure players that use VoIP.
But Gartner doesn’t paint an entirely rosy picture for VoIP. Gartner said “that despite this significant potential, conditions for the rapid expansion in the use of mobile VoIP are not yet right and are not likely to become right for at least five years and perhaps as long as eight years.”
Yeah, well when carriers like AT&T pull crap like forcing Apple to only allow VoIP over WiFi and not over a 3G data connection, it’s no wonder why mobile VoIP growth will be stunted by anti-competitive tactics. Then you have countries like Canada which outright block Skype on the iPhone. Fortunately, avid mobile phone users aren’t taking this lying down. For instance, you can jailbreak your iPhone and run VoIP over 3G no problem.
Why do we have to put up with this crippleware? I get that AT&T is a business and needs to make money. If they’re worried that flat-rate data plans that run VoIP over it will drastically hurt their voice revenue, then change your business model! Keep the flat-rate data plan, but install packet-inspection technology that detects voice packets and charge a few cents for VoIP calls. If the flat-rate data plan’s business model is outdated, which it seems to be, then change it.
Yeah, sure customers may not like the idea and certainly it seems that our culture today expects “something for nothing”, but hey, it’s business folks. I’d rather have the capability of making VoIP calls using a SIP provider or make Skype calls and pay a few pennies than not be able to use VoIP over 3G/4G at all. I’m sure if AT&T did try and charge money for VoIP calls running over a data connection that customer advocate groups, the ACLU, and people who think they should get something for nothing will run to the government and complain that AT&T is charging them money for using VoIP. Perish the thought that a business is charging you money for using their services!
Maybe we should just nationalize all the carriers like most countries have and then petition Congress to pass a law that will force the nationalized carriers to give us free mobile VoIP. Heck, with trillions of dollars being spent on the bailout package, why not free mobile VoIP at the expense of the big bad carriers? Free healthcare for everyone, free mobile VoIP for everyone, it’s all free baby! Socialism here we come!
Ok end rant. Back to Gartner…
“Mass-scale adoption of end-to-end mobile VoIP calling will not happen until fourth-generation (4G) networks are fully implemented in 2017,” said Tole Hart, research director at Gartner. “Once the basic market conditions are in place, transition to mobile portal VoIP should be fairly rapid because of the inherent convenience and end-user cost savings. In 10 years time we expect that 30 percent of mobile voice traffic will be carried out through third-party mobile portals, such as Google, Facebook, MySpace and Yahoo, which will adopt wireless VoIP service as a voice option to their current communications hub.”
A number of third parties, such as Skype, Truphone and fring, which carry VoIP traffic using a mobile phone, have cropped up in the past couple of years, offering access to voice services via Wi-Fi and/or the carriers’ wireless voice networks. This has been the most efficient way to offer the service to date because of the inconsistencies of voice services over third-generation (3G) data networks. However, with the advent of 4G networks (WiMAX and Long Term Evolution [LTE]), and increased use of smartphones with open operating systems, it is conceivable, perhaps even inevitable, that wireless voice services will be run completely over VoIP.
“Ten years from now, more than half of mobile voice traffic will be carried end-to-end using VoIP,” said Akshay Sharma, research director at Gartner. “Carriers will adopt voice services because of the increased capacity and reduced cost of delivering voice over 4G networks. Third parties will adopt a voice option for their communications hub.”
Gartner analysts warned that there will also be a number of factors that will inhibit the adoption of third-party, end-to-end VoIP services, including the delay in rolling out 4G networks because of current economic conditions and also the general plan to put 4G only in the main cities and build out from there. Nevertheless, in five to 10 years time, as 4G networks become common, mobile VoIP services will have a strong impact on the communications market.
Competing with mobile portal VoIP will be wireless carriers that offer circuit and VoIP voice and data services, and resellers and mobile virtual network operators (MVNOs) that also offer services off the carrier networks. Gartner expects this opening of the VoIP channels to spawn a number of voice services from companies that offer voice services to communities using voice as a communications link. This means that the biggest competitors to mobile VoIP may be text messaging and e-mail, as people may prefer to use these types of communication because of their non-intrusive, less emotional and less time-consuming nature.
Although the impact of the technology shift will be gradual as 4G networks roll out, Gartner advises carriers to start thinking now about how the transition will occur and how they might cooperate and partner with other types of service providers. Third-party providers, such as Google and Yahoo, should look to offer voice services today using the carriers’ networks and Wi-Fi to leverage their portfolio of services. Mobile social communities, such as Facebook and MySpace, which benefit from messaging traffic as it keeps eyeballs on their sites, should also have a voice option.
Tags: 4g, fring, gartner, long term evolution, lte, mobile voip, sip, skype, tole hard, truphone, voip, wimax
Related tags: voice services, mobile voice, voice traffic, third party, voice option, mobile
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Free Calls on Skype
3 UK is opening up its network to allow anyone with a 3 SIM and a compatible handset to have unlimited Skype-to-Skype calls and instant messages without ever having to pay. 3UK announced that on May 1st users with a 3UK SIM and a 3-provided Skype-enabled handset will be able to use Skype for free. Additionally, previous deals for free Skype required a monthly topup of £10 on pay-as-you-go and there will be no data charges. Thus, Skype will be 100% free, subject to their fair use policy of course. 
There is just a one time fee of £1.99 for a SIM card.
Anyone with a compatible unlocked 3G handset in the UK, can take advantage of free Skype calls, whether or not their phone is from 3. This is a huge barrier-busting move by 3 that changes the game. Forget about using AT&T on your iPhone who is preventing Apple from allowing Skype to run over the 3G data connection. Simply get an unlocked iPhone, stick in a 3 SIM and enjoy free Skype calls! Of course, this “free” calling deal only applies to the UK. Why do the Brits get all the fun? 
No worries, I’m sure this is just the first step before the U.S. gets in on the free Skype calling action. I wonder if there might be some synergies with the new Skype for SIP program (of which I’m a beta tester)?
According to 3 UK, “By removing these key barriers to Skype-to-Skype calls, 3 UK is creating a compelling reason for new customers to join 3 and to enjoy all the products and services available on the UK’s biggest mobile broadband network.”
According to 3 UK:
We’ve been working with Skype for over two years now and we’ve learned a lot about how our customers use the service. We know that a keen mobile Skype user is an instinctive, active communicator. They want to take full advantage of free mobile calls.Increasingly Skype use is linked to other internet communications activities, such as Facebook, Twitter and a host of other services for which our customers will happily pay a fixed fee for unlimited use.
While others have looked at Skype as a potential threat to voice and text revenues we see another advantage. Did you know when you call a friend on a different mobile network their network will charge your operator a fee for carrying the call? This is called a Mobile Termination Rate or MTR and is currently charged at around 4.7p or more for every minute of the call.
This fee is regulated by the industry regulator Ofcom but we think they’re still too high. Without these fees we’d be able to offer our customers much better value and that’s where Skype comes in. With Skype, MTRs don’t apply so we can give our customers all the minutes they like without over charging them.
At 3, we believe our customers should be able to choose how they communicate because that’s mobile as it should be; simple, useful and always good value.
Kevin Russell, Chief Executive Officer of 3 UK, said “Communication through the internet is exploding. Internet calling or VoIP, social networking, instant messaging and email are used by millions in the UK every single day. They are open to all on their PCs and laptops. We want people to be free to communicate from their mobiles in the same way as they do from their PCs.
“In future you will be able to buy a 3 SIM for unlimited Skype-to-Skype calls for less than the price of a cup of coffee and talk for as much as you want without ever paying us another penny. We won’t ask you for a top-up or a monthly commitment. If you want to talk on a mobile for free, just join us and give it a go. This is for everyone.”
Josh Silverman, President of Skype said, “Demand for mobile access from our users has never been higher. The introduction of unlimited Skype-to-Skype calls and instant messages across all 3 price plans is a really exciting move from a key partner. 3 UK clearly understands the desire for people to use Skype wherever and whenever they want. This is the first mobile network to show this kind of innovation to enable their customers to access Skype.
“We believe this is how the future looks for the Internet on mobile. With this bold move 3 UK has again shown their willingness to be the customer champion for mobile services in the UK.”
Currently, 3 UK’s growing Skype community enjoys 1.5 million minutes of free Skype-to-Skype calls every day. The launch of the first 3 Skypephone in October 2007 really kick-started the growth of free internet calling on the 3 network. With over 433 million people registered on Skype worldwide, the new free Skype-to-Skype offer from 3 opens up a world of free calling.
Two years experience of providing open access to Skype-to-Skype calling has enabled 3 and Skype to better understand the behaviour of mobile Skype users. Success with an easy-to-use Skype experience on more specialised internet-enabled handsets, such as the INQ1 and the 3 Skypephone collection; has proven to 3 that enabling customers to make free Skype calls to other Skype users on their mobiles or PCs is a real benefit.
3 UK has found that regular Skype users:
- Are less likely to churn than non-Skype users
- Use more traditional voice minutes than non-Skype users in addition to calling their Skype contacts
- Use Skype IM, but also send more SMS than non-Skype users
- Are more likely to browse the internet on their mobile
- Are higher margin customers
- Are twice as likely to access social networking sites as non-Skype customers
“Today we are moving in a clear direction towards making Skype-to-Skype calling available to all UK mobile consumers,” said Mr Russell. “We know that Skype users are instinctive communicators, keen social networkers and mobile internet users. They love the things that we are building the UK’s biggest mobile broadband network for.
“Our network is built to deliver the benefits of the internet to the mobile. That’s why we’re removing the conditions and restrictions from our current Skype offer and opening up the opportunity to try free internet calling to all UK mobile users, whether they are currently with us or a competitor network.”
Tags: 3, 3 UK, calling, free, josh silverman, kevin russell, SIM, skype, skype-to-skype, unlocked phone
Related tags: skype calls, skype skype, skype users, internet mobile, internet calling, skype
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GVDialer for Google Voice
GVdialer is an intriguing mobile application for Google Voice, supporting iPhone, BlackBerry, Android, Microsoft, and Symbian. GVdialer enables you to use Google Voice with your mobile phone while enabling some cool features. For instance, you can present your Google number as your Caller ID on outgoing mobile calls, thus keeping your mobile number private. This also gives you a one number identity to share with people.
Using the app installed you can dial directly from your phone’s contacts, speed dial, call log or keypad, and GVdialer will automatically connect the call using Google Voice.
Even cooler you have Google Voice feature access including instant access to Google Voice’s voice mail, Inbox, and GOOG-411.
As seen in the iPhone application, GVdialer lets you configure when GV would be used, i.e. on all calls, international calls, domestic calls, or ask on every call.
It costs $9.99, but definitely worth checking out
Tags: google, google voice, gvdialer, iphone, microsoft, mobile phone
Related tags: google voice, google, voice, gvdialer, mobile, calls
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OCS 2007 R2 PIC problem with AOL Fixed
Microsoft OCS 2007 R2 users were having communications issues with AOL’s AIM when federating using PIC (Public IM Connectivity) and using a Windows Server 2008 (x64) Edge role server - Windows Server 2003 (x64) is unaffected by this problem. Microsoft’s Scott Oseychik just issued a fix that solves the problem. The fix involves changing the Windows Server 2008 Edge role to initially establish the SSL dialog using the TLS_RSA_WITH_RC4_128_MD5 cipher suite.
It’s pretty easy to fix via Group Policy (gpedit.msc). Once you make the fix you should be able to successfully communicate with AOL AIM clients using Office Communicator 2007 R2 via PIC.
Click here for the resolution.
Tags: LS_RSA_WITH_RC4_128_MD5 cipher suite, Microsoft OCS 2007 R2, pic, Public IM Connectivity, Scott Oseychik
Related tags: windows server, server, using, windows, problem
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What Skype & Rich Tehrani Do in Vegas Stays in Vegas
I won’t comment on Rich taking pictures of scantily clad women unawares. In fact, I won’t comment at all. I’ll just steal Rich’s pictures of the Skype CTIA event party in Las Vegas. The glass pool table with the Skype logo looks pretty cool, as does the giant mockup of an Apple iPhone. It’s too bad they didn’t show Skype running on the iPhone though.
And what’s up with the funky dude with the mask, stilts, wearing a Skype logo? Words can’t express my thoughts, but I’ll try via the photo captions.
p.s. check out my Skype over 3G on iPhone article. You’ll enjoy it!
Not sure, but I think the ladies are holding iPhones as Rich is taking this photo. You didn’t notice? Well what were you looking at?
Get back to work Rich!
I don’t think they knew you were taking their picture. Leave the peeping Tom to me.
What the???
A guy in a cage and not one of the scantily clad women? Go figure…
I bet he has no Skype buddies
Don’t look up! He’s not wearing underwear. Ack!
Skype runs rings around the competition, so this is apropos
I could do that. No problem!
Didn’t I tell you to get back to work? Oh wait, you’re the boss. Nevermind.
Cool. Is that ice? And what are those balls?
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Tags: skype, rich tehrani, voip, apple, iphone
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AsteriskNOW 1.5.0 Released
AsteriskNOW 1.5.0, which launched as a beta in October 2008, is now available for download at http://www.asterisknow.org/downloads. Of course, existing AsteriskNOW users can simply run “yum update” to update to the latest release. I love ‘yum’ for Linux systems - it’s like Windows Update on steroids, but without the Internet Explorer GUI. 
According to AsteriskNOW, here are the notable changes since beta2:
* Updated several packages to latest versions (Asterisk, DAHDI, etc)
* Fixed more permissions issues between Asterisk and httpd/FreePBX.
* Updated to CentOS 5.3 (http://lists.centos.org/pipermail/centos-announce/2009-April/015711.html)
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Tags: asterisknowasterisknow 1.5.0, voip, asterisk, yum, linux, centos 5.3
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Digium Launches Support Services for Asterisk
Some big news from Digium. Rich Tehrani met with them yesterday to get the inside scoop. Rich takes copious notes on his iPhone, which he sent off to me to try and write up this news. Alas, I’ve been pretty busy myself, but I wanted to share Rich’s notes below, since there are some good “nuggets” in there.
For instance, from Rich’s notes I see that Switchvox 4.0 is on the verge of shipping. But the really big news is that Asterisk has announced the general availability of technical support subscriptions for open source Asterisk. Before if you wanted support from Digium, you had to purchase Asterisk Business Edition. Well, no longer. Now, all of you Asterisk fans out there that try Asterisk and get stuck can now contact Digium and get some support. No more relying on the Asterisk community to answer you questions. Not that asking the Asterisk community is a bad thing, but if you phone system is down, you can’t wait hours for someone to respond to an online posting. This could be a huge revenue-generating opportunity for Digium, which can now monetize the open source version of Asterisk with support subscriptions. I’m surprised they didn’t offer it sooner. Maybe they were afraid it would upset channel partners?
Rich’s notes:
- Open Source Subscriptions
- 2 smb subs
- And 2 enterprise class
- Incident based
- Problem: up to today needed community support or consultant with hourly rate
- Now annual sub - 3 year 10% discount
- Can call Digium based
- Level one - support local hours 12 hours - starting at your 8:00 - 7:00
- For 5 days a week
- Buys sub online
- Available in a month through the channel
- Get a key, name contact and get details when you call
- Get incident/case handled
- Can open via we or phone
- Find a bug - gets entered in bug tracker
- Gets handled like any biz edition type of bug
- Not really SLA like a commercial licensed product
- Biz edition - now only available as OEM or commercially licensed product
- They want people to buy the open source - engineering opens up 1.4 and 1.6 - first time Digium provides support for open source asterisk
- Up till now consultants, etc
- Open source - people buying business edition for support reasons
- Now getting open source subs
- Can now support enterprise class apps
- In the past - anyone who built a large network - 2 levels of enterprise class support
- 24×7 - server based
- Unlimited users
- Up to 3 names contacts
- First foray into enterprise from server side
- Up to 24×7 support
- Switchvox 4.0 on the verge of shipping
The new Asterisk support services enable companies to leverage the power of open source Asterisk with the confidence that their system will be supported by the very founders of the Asterisk movement. According to the news release, “The support subscriptions provide technical support, hardware replacements and substantial discounts on training programs to enable users to take full advantage of the power of the Asterisk platform.”
“Digium’s new subscription services give Asterisk users the best of both worlds–they can download and use Asterisk free of charge, as always, and now they can also call on Digium for technical support when needed,” said Spencer. “We think the combo of free and open, with support, is going to appeal to many of our most technical users. The Asterisk community has long been a source of great expertise through online forums, and now we’re supplementing that with the ability to call us, 24×7, for access to our Asterisk experts.”
Danny Windham, CEO of Digium, said: “As Asterisk gains traction within large businesses, demand for professional support is on the rise. Our deep knowledge of open source Asterisk and total commitment to its development makes us ideally suited to offer these new services. Companies that purchase subscriptions will receive support from the most knowledgeable group of Asterisk experts in the industry. We see this offering as a substantial step forward for Asterisk in the enterprise and a valuable service for companies of all sizes.”
Asterisk support subscriptions are bundles of services sold on an annual basis. They include technical and engineering support, consultative services, advance hardware replacement, and discounts on Asterisk training and conference passes.
Asterisk support subscriptions are available immediately from the Digium webstore at http://store.digium.com and will be available through Digium channel partners in Q2. SMB pricing begins at U.S. $595 per year for support during the subscriber’s business hours (8:00 a.m.-5:00 p.m., Monday through Friday); 24×7 support for an SMB begins at U.S. $1,995 per year. Enterprise subscriptions, including 24×7 support, begin at U.S. $3,995 per year. Pricing includes a defined number of servers supported and cases opened per year.
You can read the official news announcement here.
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Tags: technical support, subscriptions, asterisk, open source, digium, ip-pbx, linux, switchvox 4.0, voip, mark spencer, danny windham
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VoIP Defies the Bear Economy like Chuck Norris defies a 700lb Bear
An interesting report by IBISWorld, Inc. just named VoIP as the predicted #1 performer In 2009 - it even beat video games. Yet more proof that VoIP isn’t dead and indeed VoIP is the one shining bright spot in an otherwise dismal bear economy.
VoIP is indeed defying the “bears” of the economy. Reminds me of when Chuck Norris defied that huge bear in the woods. Sure, the bear got on top of Chuck and even ripped Chuck’s shirt, but this was just a training exercise for Chuck’s “You don’t want to tick me off stare.” At the end of the wresting match, Chuck just looked at the bear, the bear knew he lost, and walked away. Bears don’t mess with Chuck Norris and apparently the bears of the economy don’t mess with VoIP.
Check out the news release below after you watch Chuck give the bear his patented stare down:
IBISWorld Announces Top 10 Industries
The recession is crippling businesses across the nation, but several industries will remain unscathed by the current economic strife, according to recent Recession Updates published by industry research firm IBISWorld. As one of the nation’s most respected independent publishers of business intelligence research reports, IBISWorld today announced the top 10 industries expected to have the largest revenue growth in 2009:
INDUSTRY REVENUE GROWTH 2009
1. Voice Over Internet Protocol Providers (VoIP) 20.1%
2. ecommerce & Online Auctions 12.6%
3. Biotechnology 10.3%
4. Engine, Turbine & Power Transmission 10.0%
Equipment Manufacturing
5. Scheduled Bus Service 9.2%
6. Court Reporting Services 7.7%
7. Community Housing Services 7.5%
8. Search Engines 6.5%
9. Family Counseling 6.1%
10. Video Games 5.8%
“Emerging industries remain well represented and continue to benefit from technological innovation and cost advantages,” explained George Van Horn, senior analyst with IBISWorld. “Unfortunately, the impact of the recession is equally pronounced among sectors directly benefitting from the social and financial stress associated with the downturn.”
While only five percent of all U.S. industries are fortunate enough to be positively impacted by the recession, IBISWorld research estimates that nearly 60 percent of all industries are negatively impacted or worse.
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Tags: voip, IBISWorld, research report, chuck norris, economy, bear economy, recession
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Microsoft Roundtable is now Polycom CX5000 Unified Conference Station
Polycom and Microsoft today announced that “as part of Microsoft’s vision to broaden the availability of Microsoft RoundTable, Polycom has licensed the right to distribute RoundTable, effective April 13, 2009.” The product, renamed the Polycom CX5000 Unified Conference Station, will be available through Polycom and its channel network.
Polycom has ‘exclusive rights’, to sell the CX5000, which is a huge win for them. Although I have been a huge fan of the Microsoft Roundtable with it’s cool 360 panoramic video, my guess is that Microsoft has had difficulty selling this expensive ($4300) videoconferencing equipment.
The CX5000 when used in conjunction with Office Live Meeting service, or as part of Office Communications Server 2007, it combines content, a panoramic 360-degree view of the entire meeting room, and a separate view of the active speaker for a unique and engaging voice and video experience.
The Polycom CX5000 will be available beginning April 13, 2009, at a list price of U.S. $4,300. The CX5000 will be available in 27 countries through Polycom’s extensive channel partner network and will be available for shipment in late April. Once the Polycom CX5000 is available, RoundTable will no longer be sold. Microsoft will continue to support all RoundTable devices already sold, while Polycom will provide front-line customer support for CX5000 units sold beginning April13. To learn more about the Polycom CX5000, visit www.polycom.com/go/polycomcx5000.
You can check out my review of the Microsoft Roundtable, now called the Polycom CX5000 for more details on this product.
Tags: cx5000, microsoft roundtable, polycom cx5000 unified conference station, video conferencing, voip
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Google Voice Meet Asterisk
Nerd Vittles has another cool Asterisk recipe that combines Google Voice, voicemail transcription (via Google Voice), free calling, and of course Asterisk. Nerd does some packet sniffing and determines that Google Voice, powered by Grandcentral, is using SIP. What’s most interesting is that Nerd determine that your SIP connection and your Google Voice phone bill is only protected by a 4-digit PIN. Yikes! That’s not good.
Anyway, here’s a teaser of Nerd’s awesome recipe:
what we want to do is examine some ways to integrate the Google Voice feature set into our existing Asterisk implementations. The potential benefits are enormous. There’s free calling in the U.S., free distribution of inbound calls to multiple phone numbers scattered around the country, free SMS messaging and delivery by email, free transcription of voicemail messages into text-based emails, free conferencing, and free GOOG-411, a voice-activated service that let’s you find nearby businesses by saying where you are and what you’re looking for. For today, we’ve set our sights on the Google Voice feature set which is easiest to integrate into existing Asterisk systems: free voicemail message transcription, free calling in the United States, and free GOOG-411 directory assistance. For lack of a better term, we call it… Googlified Messaging™.
Well, what are you waiting for? Go read the entire recipe and tutorial. Great stuff!
Tags: asterisk, google, Google Voice, nerd vittles, voicemail transcription, voip
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Skype For SIP Marries Skype and IP-PBXs
Today, Skype announced it is enabling SIP-based IP-PBX to connect to the Skype network, which will allow low-cost SkypeOut calling, receiving calls from Skype users, and receiving calls from regular PSTN phone lines. Outbound calls from IP-PBX SIP handsets to Skype phones is not part of this news announcement. Skype commented it is too difficult to dial Skype usernames from a desktop handset.
Features:
- Receive and manage inbound calls from the 405 million Skype users worldwide on SIP-enabled PBX systems, connecting the company website to the PBX system using Skype click-to-call buttons
- Place calls via Skype to landlines and mobile phones worldwide from any connected SIP-enabled PBX, saving your business money with Skype’s low rates
- Purchase Skype online numbers to receive calls to the corporate PBX from landlines or mobile phones
- Manage Skype calls using your existing hardware and system applications such as call routing, conferencing, phone menus, voicemail and call recording and logging - no additional downloads or training are required
Skype For SIP is perfectly suited to businesses that already have IP-PBXs and want to connect to Skype’s network which offers low-cost calling. Skype for SIP is being launched as a closed beta program, but you can register and try to be part of the beta. Interestingly, Asterisk for Skype, a SIP-based (and IAX) open-source IP-PBX was the first to offer some tight integration with Skype.
If Skype continues to open their proprietary doors (they recently announced they were giving away the SILK codec), and now they’ve finally enabled SIP-to-Skype integration, then we can see a momentous shift from SIP trunking to Skype trunking. Customers will not only choose the lowest cost IP trunk, but also the one with the best quality. Skype is renowned for their high-quality due to their P2P architecture, so companies will look very closely at Skype trunking instead of SIP trunking.
And of course if customers shift to purchasing Skype trunks, the SIP trunks will have to follow suit, which means a win-win for businesses looking for low-cost calling.
Via Skype blog
Tags: ip-pbx, open source, sip, Skype, Skype for SIP, voip
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Dotcom-Monitor announces new SIP Monitoring tool
Today, Dotcom-Monitor announced a new SIP monitoring tool to add to its portfolio of external monitoring services. It’s similar to other web-based Monitoring-as-a-Service (MaaS) services which monitor the uptime of web servers and notify when a problem occurs. In this case, Dotcom-Monitor’s SIP Monitoring service monitors on-premise or hosted IP-PBXs.
How’s it work? Dotcom-Monitor’s SIP monitoring service makes live intermittent SIP-based calls to VoIP devices, providing real-time monitoring, alerts, and performance reports regarding SIP component connectivity. When a problem is detected, the SIP monitoring notification feature sends an alert via phone, pager, email, or SMS. Basically ,it acts as a SIP end client, placing an actual telephone call to a specified number, and checking the results of that call. The expected result of the call is setup as “Answer”, “No Answer”, “Busy”, or an Error Condition (if there is an unexpected result).
According to their representative, “real-time connectivity status reports are provided via an intuitive online Dashboard interface offering sufficient detail to help pinpoint where the error condition is occurring. This reporting functionality also includes detailed historical reports and charts for managing VoIP systems and components, including Service Level Agreement (SLA) compliance issues.”
I’m going to talk to then next week to find out more. For now, check out the news release…
Dotcom-Monitor Enhances Unified Suite of Monitoring Services with SIP Monitoring for VoIP Systems
Easy-to-Use, Cost-Effective External Service Monitors and Analyzes SIP Systems or Infrastructure for Uptime and Performance
Minneapolis, Minn. − March 18, 2009 − Dotcom-Monitor, (www.Dotcom-Monitor.com), a leading provider of externally-hosted network monitoring services, today announced the addition of a cost-saving SIP monitoring service to the company’s unified suite of monitoring capabilities. Today’s announcement adds another critical tool to Dotcom-Monitor’s portfolio of external monitoring services, which includes uptime and performance monitoring of websites, web applications, and Internet network infrastructure.
Dotcom-Monitor’s new SIP monitoring service makes live periodic SIP-based calls to VoIP devices, providing real-time monitoring, alerts, and performance reports regarding SIP component connectivity. When a problem is detected, the SIP monitoring notification feature sends an alert via phone, pager, email, or SMS. Additionally, real-time connectivity status reports are provided via an intuitive online Dashboard interface offering sufficient detail to help pinpoint where the error condition is occurring. This reporting functionality also includes detailed historical reports and charts for managing VoIP systems and components, including Service Level Agreement (SLA) compliance issues.
“Due to SLA requirements and hybrid VoIP traffic routes, it is important for VoIP monitoring to proactively mimic the end-user’s perspective from external locations, rather than only relying on passive internal network analysis systems,” said Vadim Mazo, founder and chief technical officer of Dotcom-Monitor. “Many organizations’ VoIP monitoring and uptime needs are best addressed by a simple, cost-effective external system, rather than a large, expensive in-house system. Dotcom-Monitor’s SIP monitoring service provides customers a unique, easy-to-use, targeted solution for quickly identifying and pinpointing VoIP connectivity error conditions,” noted Mazo.
The new SIP monitoring service can be configured and managed with little or no IT expertise, which is ideal for the growing number of small and mid-sized businesses (SMBs) with on-premise or hosted IP-PBXs. Its proactive monitoring ensures connectivity errors can be addressed before the errors become downtime problems for customers. Dotcom-Monitor’s SIP monitoring service ensures SMBs can rely on their VoIP systems, Service Providers can monitor their VoIP infrastructure, VoIP Wholesalers can monitor Service Provider connectivity and reliability, and VoIP VARs and managed service providers can count on client uptime and revenue.
“As the VoIP ecosystem continues to grow in scope and complexity the need for simple and affordable SIP monitoring has never been greater,” said Jonathan Fuld, CTO of SIP Print, the only provider of pure, affordable SIP call recording systems for SMBs. “In fact, SMBS and any cost-conscious organization that is dependent on SIP-based communications could benefit by investigating an externally hosted SIP monitoring provider like Dotcom-Monitor.”
Dotcom-Monitor’s SIP Monitoring is available immediately by visiting: www.dotcom-monitor.com
Tags: Dotcom-Monitor, phone, SIP Monitoring, sla, sms, voip
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How to make OCS 2007 R2 non-RFC 3966-compliant using RemovePlusFromRequestURI
Office Communications Server 2007 Mediation Server uses a plus sign (+) to prefix E.164 numbers in the Request Uniform Resource Identifier (URI) for outgoing calls. Unfortunately, some IP-PBXs don’t comply with RFC 3966 and do not accept numbers that are prefixed with a plus sign (+).
As the UCSpotting blog points out:
To make sure that OCS 2007 operates correctly with non-RFC 3966-compliant PBXs, Microsoft released an update for Mediation Server (R1), which is described in KB articles 952780 and 952785. After installing the update, it’s necessary to create a configuration file - MediationServerSvc.exe.config - with the following content:
"1.0" encoding=“utf-8″ ?> <configuration> <appSettings> <add key=“RemovePlusFromRequestURI” value=“Yes” /> </appSettings> </configuration>
In OCS 2007 R2, Microsoft changed this slightly negating the need for the above configuration file. There’s a new WMI setting, RemovePlusFromRequestURI , which is described in this TechNet article called Enterprise Voice Server-Side Components.
According to the TechNet article, Office Communications Server 2007 R2 introduces two new Windows Management Instrumentation (WMI) settings for Mediation Server. The first new setting specifies how Mediation Server processes E.164 numbers in outbound calls. The second new setting enables Quality of Service (QoS) marking on Mediation Server.
Handling E.164 Numbers in Outbound Calls (OCS 2007 R2)
By default, E.164 numbers in the Request Uniform Resource Identifier (URI) for outgoing calls are prefixed with a plus sign (+). Most Private Branch eXchanges (PBXs) process such numbers without problem. Certain PBXs, however, do not accept numbers that are prefixed with a plus sign.
To ensure interoperability with these PBXs, Mediation Server has a new WMI Boolean setting called RemovePlusFromRequestURI, which has two values: TRUE and FALSE. If your PBX does not accept numbers prefixed with a plus sign, the value for the WMI setting should be set to TRUE, which causes Mediation Server to strip the plus sign from a Request URI for outbound calls. The default is FALSE, which causes Mediation Server to pass the outgoing INVITE’s Request URI, To URI, and From URI unchanged.
The TechNet article also discusses compatibility with PBXs that do not support the plus (+) sign.
By default, E.164 numbers in the Request URI of outgoing calls from Office Communications Server 2007 R2 are prefixed with a plus sign. Most PBXs process such numbers without problem. Some PBXs, however, do not accept numbers that are prefixed with a plus sign and do not route those calls correctly.
Additionally, the From headers of inbound calls from some PBXs does not conform to RFC 3966 because they are not prefixed with a plus sign. Microsoft Office Communicator cannot resolve these numbers to the correct user.
To assure interoperability with these PBXs, Office Communications Server 2007 R2 has a new Mediation Server setting for WMI called RemovePlusFromRequestURI. This setting can be set to YES or NO. The default value is NO.
- If a PBX downstream from the Office Communications Server 2007 R2 Mediation server does not accept numbers prefixed with a plus sign, set the value of RemovePlusFromRequestURI to YES. This causes Mediation Server to remove the plus signs from the Request URIs of outgoing calls. It also causes the plus signs to be removed from the To and From URIs.
- If the downstream PBX accepts numbers prefixed with plus signs, leave the value of RemovePlusFromRequestURI set to its default value of NO. This causes Office Communications Server 2007 Mediation Server to pass Request URIs, To URIs, and From URIs unchanged (that is, with plus signs).
UCSpotting’s excellent article explains all this, and includes a nice VBscript for how to change the boolean value (true or false). Check it out!
Tags: Microsoft OCS 2007 R2, OCS 2007 R2, RemovePlusFromRequestURI, e.164, outbound calling, plus sign, ucspotting, unified communications
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- Windows Server 2008 RDS Does VoIP - Mar 11, 2009
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Microsoft OCS 2010 Will Finally Eliminate the PBX
Well, Microsoft has let the cat out of the bag and leaked word that Microsoft OCS 2010 will “remove the need for PBX equipment within your organization”. I’m certainly not surprised. Let’s flash back to last year where I wrote and article titled Microsoft OCS 2007 R2 Heralds the Death of the IP-PBX. In it I wrote:
“Office Communications Server 2007 R2, debuting just one year after the Microsoft unified communications launch, highlights the pace of innovation that is possible with software,” said Stephen Elop, president of the Microsoft Business Division at Microsoft. “This new release puts Microsoft on a rapid path to deliver voice software that does much more than a network private branch exchange (PBX) and with much less cost.”
Interesting quote, eh? Does this not sound like Microsoft is sounding the death knell for the network PBX (IP-PBX)? This is an interesting turn of events. Microsoft hasn’t been pitching OCS 2007 as an IP-PBX replacement, but rather as something complementary. In fact, I remember talking with Microsoft about this last year (2007) and they went out of their way to explain that OCS 2007 is not an IP-PBX replacement. Also, Microsoft has many IP-PBX partners in the OCS 2007 arena, including Mitel, Nortel, and others. Slip of the tongue? Or is Microsoft going full-out into the IP-PBX arena? Certainly, the fear by many IP-PBX vendors is that one day Microsoft will offer a full-fledged software-based IP-PBX replacement, but I don’t think that day has come yet - even with the new features in OCS 2007 R2.
Now with OCS 2007 R2 fully launched and with added support for direct SIP trunking, the next logical step is a 100% Microsoft UC solution without the need for a PBX/IP-PBX at all. Of course, Microsoft OCS 2007 R2 is still currently very limited in the support it has for SIP IP phones. Most businesses aren’t ready to toss desktop phones for a 100% software-based softphone solution, i.e. Microsoft Communicator. So OCS 2010 will have to support SIP phones from popular SIP phone players such as Aastra, Polycom, and snom. Perhaps Microsoft will borrow or acquire the technology from SmartSIP, which recently launched an add-on for OCS 2007 R2 that enables any SIP phone to work with OCS.
So where did I hear that Microsoft was aiming to eliminate the need for a PBX in OCS? I discovered the information within a document on Microsoft’s website titled ‘Microsoft Unified Communications Business Value Tool’. On Page 24 it states:
You will deploy Office Communications Server 2010, which expands on the communications capabilities delivered in OCS 2007 R2. This release is designed to remove the need for PBX equipment within your organization and replace it with an integrated communications system that dramatically reduces management costs and gives end users innovative tools to communicate and collaborate across geographic boundaries from their office, home or on the road.
Not only do they state they will eliminate the PBX, but they declare the next version name of OCS (OCS 2010), which as far as I know Microsoft hadn’t announced yet. Many UC/VoIP experts predicted that eventually Microsoft would attack the IP-PBX space alone, but one has to wonder if alienating their IP-PBX partners is such a good idea. One of their strongest OCS partners is Nortel, who is experiencing financial difficulties and is probably not in a position to pressure Microsoft to back off. Mitel is another strong partner as well that could be impacted by Microsoft’s decision. Of course, Nortel and Mitel could still go after the SIP-based IP phone space within the OCS arena, but the IP phone market is much more of a commodity with a much lower margin than a full-fledged IP-PBX. Of course, there’s always the high-end media phone market with large margins. For instance, Polycom recently announced their VVX1500 media phone, which created some buzz.
I doubt OCS 2010 will have all the advanced call center functionality you get from Nortel, Avaya, Mitel, etc. After all, this will be Microsoft’s first release that doesn’t rely on the IP-PBX to do the intelligent call routing & handling. They’ll probably have some rudimentary call queues and skills-based routing, but not much else. Don’t expect predictive dialing in OCS 2010, a mainstay of the call center market. Still, a 100% software-based IP-PBX with unified communications capabilities will be a compelling choice for many businesses.
Tags: microsoft, Microsoft OCS, Microsoft OCS 2007 R2, mitel, nortel, sip, SmartSIP, unified communications, voip
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Windows Server 2008 RDS Does VoIP
Terminal Services allows you to remotely run applications as well as perform remote administrative duties on servers. It has allowed remote audio to be streamed over IP from the remote computer to your local computer (audio redirection) but has never allowed the microphone or line-in port to be redirected. If Microsoft did, you could do VoIP. Of course, you’d have to redirect from the local PC to the remote server and not the other way around. Well read on…
Continue reading Windows Server 2008 RDS Does VoIP…
Tags: audio recording, audio redirection, microphone, microsoft, skype, Terminal Services, unified communications, vdi, Virtual Desktop Infrastructure, voip, Windows Server 2008 RDS
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Polycom VVX 1500 Media Phone Game Changer?
Today, Polycom has launched the Polycom VVX 1500 touch-screen business media phone, a new VoIP phone that combines IP telephony with business-class video and the ability to integrate with business applications. Recently, Verizon make a big splash with their consumer-class Verizon Hub, a multimedia phone that combines VoIP, Internet access, color screen, video streaming, and more. One could easily make the case that the Polycom VVX 1500 is the “business-class” version of the consumer-oriented Verizon Hub phone.
Although there are many similar features and both could be classified as “media phones“, the Verizon Hub does not do video conferencing, since it does not have an embedded camera. The Polycom VVX 1500 on the other hand does have a video camera embedded and is therefore more suited to video conferencing, which is more prevalent in the business world any way.
The Polycom VVX 1500 combines a personal video conferencing system with a fully featured voice over IP (VoIP) telephone along with Polycom HD Voice (wideband telephony) and an open application programming interface (API) and microbrowser for real-time delivery of personalized Web content. It also includes a color touch-screen interface making this a very unique business IP phone.
So is a business-class media phone with a color touch-screen, web browsing, and video conferencing capabilities a game changer in the VoIP space? Well, the VVX 1500 has a list price of U.S. $1,099, so this is not an IP phone for everyone’s desk in a corporate office. A decent IP phone for the every day worker can be had for $150-$300 which is much less expensive. However, for business executives, CEOs, VPs, and other high-level management, the VVX 1500 is a very attractive IP phone. Often times if a VP or CEO has to have a high-quality video conference, they have to reserve a high-quality video conferencing system located in a particular boardroom. With the VVX 1500 they can stay at their desk and have their meeting. Further, impromptu video conferencing with co-workers sporting a VVX 1500 on their desk can be had allowing for quick collaborative meetings.
In-Stat is very high on the prospects for business-class media phones. According to Keith Nissen, principal analyst at In-Stat, “We anticipate that within five years, nearly 10 million business media phones will be shipped worldwide, generating more than U.S. $3 billion in annual revenue. They are a key to the future of the IP PBX business.” He added, “With its rich heritage in voice and visual communications and content sharing, Polycom is well positioned to be a leader in this new world of communications. The company’s VVX 1500 is the first business media phone that enables customers to work more efficiently and effectively than ever before by tying together voice and visual communication with critical business processes.”
Polycom VVX 1500 Touch-Screen
“There is growing demand from our service providers and customers to help them configure video within our BroadWorks call control platform,” said Mike Tessler, CEO of BroadSoft. “We have a long history of teaming with Polycom to deliver high quality hosted VoIP solutions, and the VVX 1500 is especially compelling because it goes far beyond the functionality of a traditional video phone by combining rich telephony, business-class video and an applications platform that is all deeply integrated with the BroadWorks platform, and it is extremely easy-to-use.”
The VVX 1500 was also specifically designed for lower power consumption, using power over Ethernet (PoE) using IEEE 802.3af, and requiring less than half the power of similar competing products such as traditional video phones. The device’s cool smart-motion technology enables the screen to go into power-save mode when no one is in the office.
The VVX 1500 features an open API and microbrowser that enable third-party application developers to integrate VVX 1500 with business applications such as unified communications, customer relationship management (CRM), and appointment management systems. The always-on, touch-screen user interface of the VVX 1500 includes a menu screen on which developers can place icons for users to locate and start their applications.
Polycom VVX 1500 Profile View
The VVX 1500 comes bundled with several applications including the Polycom Productivity Suite, which enables users to initiate and control audio conference calls right from the device’s screen as well as record calls locally using a flash drive in the phone’s USB port. The VVX 1500 also features a free Web service called My Info Portal through which customers can select to receive content such as local weather reports and other personalized information on the screen when the device is not in a voice or video call.
Interoperability is not a problem since the VVX 1500 uses the same Session Initiation Protocol (SIP) software as incorporated in Polycom’s SoundPoint IP and SoundStation IP desktop and conference phone product lines to communicate with SIP based IP-PBXs and hosted SIP servers. The product is in the process of being SIP video-certified by Polycom’s ecosystem of more than 30 VIP and VoIP Field Verified call control partners, including BroadSoft, Deltapath, NEC Sphere, Objectworld, and Zultys.
“Our customers consistently seek better leverage of their communication systems to improve productivity and reduce costs. They also expect Polycom to continuously deliver innovative, intuitive products to market,” said Sunil Bhalla, senior vice president and general manager of Voice Communications Solutions at Polycom. “Our leadership and legacy in both voice and video communications enables us to develop a truly unique device. The VVX 1500 is the business media phone to combine a superior business-grade VoIP telephone that features our renowned HD Voice with one-touch video and access to key enterprise applications. We’re delighted propel collaborative communications to the next level with this ground-breaking device.”
The Polycom VVX 1500 will be available this month through Polycom’s channel partner network at a list price of U.S. $1,099. To learn more about the Polycom VVX 1500, visit www.polycom.com/vvx1500.
Tags: broadsoft, ip phone, ip-pbx, keith nissen, mike tessler, polycom, sip, SoundPoint IP, SoundStation IP, sunil bhalla, video, video conferencing, voice, voip, VVX 1500
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- EZCallerID.com Hosted CNAM for Enhanced Caller-ID on any IP-PBX Launches - Feb 25, 2009
- Dual Stack SIP and Skype IP Phone Coming - Jan 30, 2009
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UCSniff2.0 launches
Greg Galitzine has the goods on a new IP video sniffer/recording product called UCSniff2.0.
According to Greg, “Until now, the information has only been posted on security boards and community sites, and on the SourceForge site at http://ucsniff.sourceforge.net/“. And of course, as you probably surmised, since it’s on SourceForge, it’s “free” under a GPL license.
According to their website:
UCSniff is an exciting new VoIP Security Assessment tool that leverages existing open source software into several useful features, allowing VoIP owners and security professionals to rapidly test for the threat of unauthorized VoIP and Video Eavesdropping. Written in C, and initially released for Linux systems, the software is freely available for anyone to download, under the GPLv3 license. Some useful features of UCSniff that have been combined together into a single package:
- Allows targeting of VoIP Users based on Corporate Directory and/or extensions
- Support for automatically recording private IP video conversations
- Automatically re-creates and saves entire voice conversations to a single file that can be played back by media players
- Support for G.722 and G.711 u-law compression codecs
- Support for H.264 Video codec
- Automated VLAN Hop and Discovery support
- A UC Sniffer (VoIP and Video) combined with a MitM re-direction tool
- Monitor Mode
- Sniffs entire conversation if only one phone is in source VLAN
Read Greg’s trip report from Dallas for more info.
Tags: G.711, G.722, greg galitzine, H.263, IP Video sniffer, sipera, UCSniff, UCSniff2.0, video recording, voip
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- Digium Headquarters Tour - Aug 26, 2008
- Truphone on iPhone Apple Store enables VoIP - Jul 11, 2008
- Flash-to-SIP Video Calls - May 27, 2008
- Shoretel Rumors - May 01, 2008
- TMC’s Greg Galitzine Interviews Dialogic’s Jeff Dworkin - Jan 16, 2008
- Communications Developer Conference Packed with Developers - May 15, 2007
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SmartSIP Launches for OCS 2007 R2 Enabling Any SIP Phone & Any SIP Trunking Service Provider
OCS 2007 R2 won’t replace your PBX just yet. However, their latest R2 version adds the ability to do direct SIP trunking, thus bypassing the need for an IP-PBX.
One drawback however is that Microsoft only supports direct SIP trunking with two providers, namely Global Crossing and Sprint. Well that’s pretty lame, considering their are dozens of decent SIP trunking service providers and probably hundreds across the entire world.
Fortunately, Mike Stacy an OCS 2007 guru, over at Evangelyze Communications has some products that enhance OCS 2007 R2 functionality. One such product is SmartSIP which launches tomorrow. According to Mike, the first dot release due next month will add the capability to use standard SIP phones with OCS. Currently, you have limited options namely Tanjay or Snom phones, but with SmartSIP you can use a Polycom IP phone, an Aastra IP phone, or dare I say, a Cisco IP phone connected to OCS 2007 R2. With the Cisco SIP firmware load of course.
Tags: aastra, cisco, Evangelyze Communications, ip phone, microsoft, Microsoft OCS 2007 R3, mike stacy, ocs 2007 r2, polycom, service provider, sip, SIP trunking, smartsip, unified communications, voip
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- Microsoft Office Communication Server R2 ships in December plus Hosted OCS Coming - Sep 23, 2008
- Objectworld Offers Unified Communications for 7 Cents a Day - Jul 22, 2008
- Microsoft Drops VoIP from Windows Live Messenger - Jun 26, 2008
- Hmmm… Buy an IP-PBX at Costco - May 29, 2008
- Cisco and Microsoft in Clash of the UC Titans - Apr 24, 2008
- Microsoft Response Point Announces SIP Trunking Service Providers - Apr 18, 2008
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SpinVox Transcribes Skype Voicemail & VoiceScribe does the same for Asterisk
Skype users can now have their voicemails converted into text via SpinVox. Today, SpinVox announced that your Skype voicemails transcribed and sent to you via SMS for €0.20/£0.17/25 cents plus the cost of the SMS. SimulScribe, now PhoneTag, is a similar service, that Rich Tehrani uses regularly. GotVoice is yet another one.
But how about another cool TTS app that is currently ‘free’ and works with the popular open source Asterisk platform? VoiceScribe is a beta web-service for Asterisk that converts your voicemail to text and delivers them to you via e-mail. What’s cool about this is how easy it is to integrate with Asterisk, trixbox CE, and trixbox Pro. I tested it with trixbox Pro and it worked flawlessly in just minutes. It uses the Nuance engine. The accuracy was OK, but I’m told by VoiceScribe’s Mitchel Constantin, “Quality will get much better.”
Simply edit /etc/asterisk/voicemail.conf, go to the [general] section and make sure wav49 is the default format. Also add a line with mailcmd that sends an email with your voicemail attachment to their hosted servers.
Here’s a sample of the 4 lines you need in voicemail.conf:
Continue reading SpinVox Transcribes Skype Voicemail & VoiceScribe does the same for Asterisk…
Tags: asterisk, Mitchel Constantin, skype, spinvox, text-to-speech, trixbox ce, trixbox pro, tts, voicemail
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- PIKA for Asterisk boards now trixbox CE compatible - Jan 22, 2008
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Karaka Bridges XMPP and Skype
Vipadia announced the release under the GPLv2 of Karaka, the open-source XMPP-Skype Gateway which connects the XMPP and Skype networks.
Karaka is a scalable distributed XMPP transport that bridges instant messaging and presence between a user’s XMPP and Skype accounts. This will for instance enable Skype-to-Google Talk instant messaging. In theory AOL’s AIM should work, since I believe they also support XMPP. In addition to full presence and instant messaging exchange, it also supports multi-user chat (”conference rooms”). Karaka implements the XMPP standards XEP-0100 for gateway support, XEP-0045 for multi-user chats and XEP-0144 for roster exchange.
According to Vipadia, “Existing Skype interconnect solutions focus on bridging voice even though the primary use of Skype is for instant messaging and associated presence data. Interconnecting with Skype messaging and presence has been a major stumbling block for many who wish to offer Skype interconnection to their network. Karaka bridges the XMPP and Skype clouds, removing this stumbling block by converting Skype messaging and presence to the popular XMPP protocol as used by, e.g., Google Talk.”
Karaka is licensed under the GPLv2 and is hosted on Google Code at http://code.google.com/p/karaka/.
Check out out @ http://vipadia.com/products/karaka/.
Tags: google talk, im, karaka, presence, skype, vipadia, voip, xmpp
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Cisco TelePresence Video Conferencing Enables Fans to Interact with NBA Players
This weekend, the NBA and ESPN will utilize Cisco TelePresence technology to create an interactive “in-person” forum to allow athletes and fans to interact using Cisco’s videoconferencing / telepresence technology.
According to Cisco, Cisco TelePresence has been deployed on the All-Star Jam Session floor allowing fans to interact “face-to-face” with NBA stars from the convention center’s green room and backstage at the arena. On-site Cisco digital signage will provide All-Star programming content and live updates of the NBA Jam Session Trivia contest.
Additionally, ESPN will broadcast exclusive NBA All-Star Weekend reports on the network’s popular news shows directly from the Cisco TelePresence system in Phoenix back to ESPN’s studios in Bristol, Connecticut.
Here’s the full details:
ESPN and NBA Bring Fans Closer to All-Star Game Action With Cisco TelePresence Technology
Cisco TelePresence Technology Will Be Used to Create an Interactive Experience Between Fans and NBA Players and Legends at NBA All-Star Weekend
NBA All-Star Weekend — The National Basketball Association (NBA) and ESPN today announced that they will utilize Cisco (NASDAQ: CSCO) TelePresence(TM) technology to create an immersive “in-person” experience that will bring NBA players and legends closer to fans at the NBA All-Star Game in Phoenix this weekend.
Facts:
• ESPN will broadcast exclusive NBA All-Star Weekend reports on the network’s news and information shows, such as “SportsCenter,” directly from the Cisco TelePresence system in Phoenix back to ESPN’s studio in Bristol, Conn.
• With the implementation of TelePresence in ESPN broadcast operations, the sports network will be able to deliver a higher level of interactive sports broadcasting for major events happening across the globe.
• During NBA All-Star, Cisco will deploy two Cisco TelePresence units at the Phoenix Convention Center and one at U.S. Airways Center for the All-Star Game activities. One of the Cisco TelePresence systems will be located directly on the show floor at the All-Star Jam Session allowing fans to interact “face-to-face” with NBA stars from the Convention Center green room and backstage at the arena.
• Fans will be able to participate in trivia contests and interact with current NBA players and legends, such as the Oklahoma City Thunder’s Kevin Durant, Toronto Raptors’ Jason Kapono and former Phoenix Suns’ legend Dan Majerle, throughout the weekend via the Cisco TelePresence systems at NBA Jam Session.
• In addition to TelePresence, the NBA will utilize the Cisco® wireless press center for real-time mobile media reporting on multiple All-Star events happening in Phoenix. And as an official technology partner of the NBA, Cisco digital signage, part of the Cisco Digital Media System, will be on-site to provide All-Star programming content and live updates of the NBA Jam Session Trivia contest.
Steve Hellmuth, executive vice president of technology and operations, NBA
“Cisco has been helping the NBA stay connected since 2007 and we’re thrilled to extend this interaction to our fans by bringing Cisco TelePresence to the Jam Session show floor. We will be able to provide our fans with unprecedented access to both current NBA players and legends.”
Ed Erhardt, president of customer marketing and sales, ESPN
“ESPN is working with Cisco to enhance the NBA All-Star viewing experience for basketball fans. Thanks to Cisco, we’ll be able to offer our audience a closer look at the All-Star Game with exclusive interviews via Cisco TelePresence.”
Alan Cohen, vice president, enterprise marketing, Cisco “By deploying Cisco TelePresence directly on the NBA Jam Session show floor, we are bridging the gap between athletes and fans, giving basketball enthusiasts a virtual pass to the arena with a unique ability to interact with players in new ways that were not possible at previous marquee sporting events. Building on our relationship with the NBA, we’re bringing fans closer to the game, from the boardroom to the locker room.”
Related Links:
- Cisco Sports Web Site
- NBA All-Star Weekend Web Site
- ESPN.com NBA All-Star coverage
- Cisco Video - Collaboration and Innovation: Bringing Fans Closer to the Game
- Cisco Video - Houston Rockets Center Yao Ming Using Cisco TelePresence to Field Press Inquiries
- Cisco Videos - The Human Network Effect in Sports
- Cisco Photos - Phoenix Suns Forward Grant Hill Reaches Global Fans with Cisco TelePresence
Tags: cisco, cisco telepresence, espn, nba, nba all-star weekend, telepresence, video conferencing
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31 Million IP Phones shipped by Mayan 2012 Doomsday, the Media Phone, & Slow Consumer Adoption
According to In-stat, nearly 31 Million Business IP Phones will ship in 2012. That’s if the Mayan 2012 Doomsday Prophecy doesn’t come to pass. You know, the one where the Mayan calendar ends on December 21st, 2012 - the same date as the Winter Solstice and when the Earth will be in galactic alignment with the massive black hole at the center of our galaxy, as well as our sun, resulting in a planetary shift. The date 12-21-12 reads as, A-B-B-A-A-B. Since the Hebrew language is read from right to left, this date would read BA ABBA. The Hebrew translation for BA ABBA is “Father comes” or “Father is coming”. If you believe in that sort of thing. Here’s a History Channel clip about 2012 that might bring out the conspiracy/doomsday nut inside you.
And then there is a Sony Pictures movie titled 2012 coming out this summer, as seen by this trailer:
Anyway, In-stat explains, “Within the business market, corded IP phones remain the standard, and will continue to dominate the enterprise IP phone market through 2012,” says Norm Bogen, In-Stat analyst. “However, WLAN and IP DECT phones continue to grow, especially within some specific vertical and geographical markets.”
Recent research by In-Stat found the following:
- Cisco, Avaya, and Nortel are leading the market for enterprise IP phones.
- Wi-Fi integration in cellular phones is growing rapidly; however, the majority of Wi-Fi/cellular phones are not designed for VoIP.
- Uniden holds top market share for consumer IP corded phones
The In-stat report points out that the IP phone market is “a tale of two markets” with IP phones thriving in business but as for the home consumer, not so much. By 2012, 31 million voice-centric business IP phones will ship but the consumer side will be outpaced by businesses more than 10 to 1. Why the slow consumer adoption of IP phones?
According to In-stat, “the nascent consumer market for voice-centric IP phones is being subjugated by the introduction of IP media phones, such as the Verizon Hub and AT&T HomeManager that support both IP communications, as well as delivery of Internet information and multimedia content.”
I think they’re a little premature in that statement. The Verizon Hub is a cool device, but it literally just came out, so it is not affecting consumers from buying IP phones at home. I think the reason is that consumers are happy with their home cordless phones with built-in answering machines. Some even have the multi-handset cordless phone systems, which allow you to strategically place handsets around the house with the ability to screen callers and remotely check the answering machine without going to the base unit. There just isn’t a good reason to purchase a $150-$300 corded/desktop IP phone for the home.
You could argue that a Wi-Fi phone might be a good option for the consumer. Wi-Fi phones are less expensive than desktop IP phones, they’re mobile, and they can get you cheaper or even free calling (i.e. Skype-to-Skype calls). But Wi-Fi phones have notoriously bad battery life. A better option in my opinion is a 
DECT 6.0 phone device with VoIP capabilities, such as the Philips VOIP841 Skype phone. Although there are other WiFi and DECT phones worth a look. I should point out that Wi-Fi phones have the advantage over DECT of sometimes offering a built-in browser so you can access the web.
Let’s look at how In-stat defines “media phone” and “why the media phone”:
The media phone is a new category of broadband device that combines the power of the PC with the performance of a telephone. The result is an always-on multimedia broadband device that is perfect for accessing online news and weather, viewing videos, and a host of other applications. In-Stat believes that the media phone will complement the PC, TV, and mobile handset, becoming an indispensable 4th screen in the home. Service providers and IP PBX vendors, alike, are introducing media phones because they add value to traditional voice telephones and related services.
Verizon Hub, a sample media phone
I certainly agree that consumers will start to adopt “media phones” in their homes, but only if the media phones are subsidized by the carrier. They’ll be too expensive otherwise. It worked for the cell phone market, so it can work for the home as well. There have been plenty of times I wanted to check the weather or current movie times, but had to boot up my PC in order to look up information. Having a media phone in the living room with instant Internet access is a nice feature to have. I do agree with the In-stat report that businesses will continue to be the main driver behind IP phone sales, but I wouldn’t be surprised to see traditional phone manufacturers such as Uniden developing cordless IP phones for the home market that offer Internet access.
You can download a free copy of In-Stat’s media phone research report: The Media Phone Has Arrived!
Relatedly, the research, “IP Phones Worldwide-On the Desk and Beyond” covers the worldwide market for voice-centric IP phones. It includes:
- IP phone vendor market shares for 2007 and 1H2008, segmented by phone type and consumer versus business
- A 5-year forecast by IP phone type (Corded, WLAN, Cordless DECT, Dual-mode Cellular/WLAN, Consumer, Business)
- Analysis of trends in business and consumer markets
- Profiles of more than two dozen vendors
Tags: 2012, DECT, doomsday, in-stat, ip phones, mayan, media phone, research report, skype, Verizon Hub, voip, wi-fi phone
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Philadelphia City Council approves a Verizon FiOS franchise agreement
The Philadelphia City Council voted unanimously to approve a Verizon FiOS franchise agreement that will allow Verizon to offer fiber-based digital TV, voice, and Internet to Philly’s 600,000 residents. Verizon will spend $1 billion to build out the network which will offer HDTV channels, voice, and high-speed Internet.
City Council members said the deal was “not perfect” and they tried to push for more minority contractor involvement. Now how is it that a City Council has the right to demand who a company hires? There are already state and federal anti-discrimination laws on the books, so why the hell do companies have to answer to some local city council mafia? Obviously, Verizon wants the business and they have to suck-up to this local city council if they want to be granted the franchise.
The vote taken yesterday gave Verizon a 15-year franchise agreement. What will the Philadelphia City Council demand from Verizon in 15 years when it’s time to renew? It’s not like Verizon is going to say ‘no’ to any demands after shelling out $1 billion to build the fiber network. Why do we even have TV franchise agreements any more? Haven’t they outlived their usefulness? It’s such a scam by local governments to make cable companies - and now carriers to “pay the man” if they want to do business in the town. Phone companies don’t have to pay local franchise fees to provide telephone service, so why do TV providers have to pay? It’s a legalized mafia racket if you ask me.
Philadelphia residents and the local government should be ecstatic that Verizon is targeting the city first before many other major cities. They should be grateful.
Tags: FiOS, Philadelphia City Council, TV, Verizon
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TANDBERG PrecisionHD USB Camera Sports 720p HD Video
TANDBERG’s new PrecisionHD USB Camera is the first webcam to offer HD video at 720p with 30 frames per second. It is optimized for Microsoft Office Communications Server 2007 Release 2 providing business-quality HD video communication. It also includes a built-in noise canceling microphone and automatic focus.
The webcam can be hooked onto a laptop (as shown above), making this an excellent mobile videoconferencing choice. It’s worth noting that OCS 2007 R2 not only supports HD720p (1280×720 1.5Mbps), but it also now supports an “unlimited” video bitrate setting as seen here:
Now we just need someone to build a 1080p webcam! Of course, the bandwidth required for that might be too much to be practical.
Tags: 1080p, 720p, Camera, HD Video, Microsoft Office Communications Server 2007 Release 2, PrecisionHD, TANDBERG, USB, video conferencing
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Cisco Unified MeetingPlace 7.0 with WebEx integration Review
Here is a preview of the TMC Labs review of Cisco Unified MeetingPlace 7.0, which is scheduled to be published in the March issue of Internet Telephony Magazine. Enjoy!
Cisco Systems, Inc.
Web site: http://www.cisco.com
Pricing:
40 concurrent user licenses (750 users), audio conferencing, includes application server software and media server hardware for voice conferencing - $79,999 ($2000/concurrent user)
100 concurrent user licenses (2000 users), audio conferencing, includes application server software and media server hardware for voice conferencing - $127,999 ($1280/concurrent user)
In both sample pricing scenarios, it also includes 6 concurrent web and 6 concurrent video licenses (customer needs to purchase video blade to use video licenses)
Note: As seen by the two example prices the concurrent user price comes down quickly as the system size grows.
Additionally, the customer would also need to purchase a $12,000 Cisco Media Convergence Servers, which is a standard server running Linux
//////////////////////////////////
RATINGS (0-5)
Installation: 5
Documentation: Not tested
Features: 5
GUI: 5
Overall: A
///////////////////////////////////
Unified communications is all the rage these days, especially when businesses are looking to optimize productivity and reduce costs. But what really is “unified communications”? You might be surprised at the different answers you’d get from various people. Many would simply say it is voice, video, email, and data (Web) collaboration with some business processes or rules applied. At a high-level this is an accurate assessment, but not all implementations of unified communications are the same nor do they all encompass the same communication mediums. Further, some UC systems only work with fellow co-workers and therefore are an island onto themselves when dealing with customers who cannot participate in the unified communications platform for collaborative meetings.
Continue reading Cisco Unified MeetingPlace 7.0 with WebEx integration Review…
Tags: cisco, Cisco Unified MeetingPlace 7.0, collaboration, review, unified communications, video, voip, webex
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ITEXPO says “Slumping economy? What slumping economy?” - VoIP tradeshow thrives
Many people I talked to at ITEXPO before the exhibit floor opened had muted expectations about the attendance, because of the slumping economy. But I knew better… I talked to Rich Tehrani and he told me a week before the show opened that the pre-registration numbers were up from last year. VoIP when properly deployed and with the right product has an extremely fast ROI and can result in cost savings in just a short period of time. So I wasn’t worried that ITEXPO would be a resounding success.
But the VoIP industry is not without its curmudgeons. Alec Sauders wrote a scathing post titled “2008: The Year VoIP died”, which sparked a fierce debate amongst VoIP industry insiders. He wrote:
VoIP events are suffocating too. VON was a spectacular flameout, despite the best efforts of Jeff Pulver and his band of merry men to transform it from a voice only show into a voice, video and more show. At least the Pulverites understood where the future was, even if unable to craft a profitable event around those varied interests. There’ll be more of the same next year, I fear. Initial reports from this fall were that VoiceCon was an understated and quiet affair. Lawn bowling anyone?
Well, ITEXPO definitely didn’t have any lawn bowling. If you tried to chuck a bowling ball down the aisle, it wouldn’t travel 5 feet before clipping someone’s ankles. ITEXPO has solid traffic the entire show, which is a rare feat for any tradeshow these days. ITEXPO isn’t VON and it isn’t VoiceCon, and probably shouldn’t be compared to either. One thing everyone could agree upon when it comes to ITEXPO is that this show brings “buyers” to the show. It’s why exhibitors keep coming back. Exhibitors actually generate “leads” and close deals at ITEXPO where as VON (R.I.P.) and other IP communications shows were more about networking and trying to strike partnerships. ITEXPO has evolved over the years into a “networking” event as well, but its core strength has always been that it brings buyers, resellers, distributors, etc. to the show.
I found many positive ITEXPO reports from industry analysts, media, bloggers, and VoIP vendors on the Web that talked about just how successful ITEXPO was.
Here are a few of them:
Considering Convergence - Network World by Matthew Nickasch:
The conference events concluded Wednesday afternoon at Internet Telephony EXPO (ITEXPO) East 2009 in Miami Beach, FL. With a very high attendee and exhibitor count, it is clear that VoIP, convergence, and UC technologies are at the forefront of interest and innovation.
Community: ITEXPO is a wonderful example of the strong industry community that exists within the IP communications market. Highly-competing developers, integrators, and VARs maximize on the opportunity to discuss the state of the industry, exciting new technological developments, and tools for delivering better and more innovative customer solutions.
Greg Galitzine, Group Editorial Director for TMCnet.com - VoIP is Alive and Well
I’ve just returned from ITEXPO East 2009, which was held in Miami Beach, February 2-4. One of the trends that emerged at the show is that the VoIP industry is indeed alive and well. The show was well attended by people looking for solutions, the Exhibit Hall was full of vendors who were only too happy to speak to the attendees, the meeting rooms were full, the conference sessions were well attended, and all in all the show served as a barometer for the industry.
Joe Staples, Interactive Intelligence Internet Telephony Expo Wrap-Up
There’s an economic slowdown? Somebody must have forgotten to tell the show attendees. Traffic seemed as strong as ever (show promoters reference a 30% increase in attendance). The conference sessions were well attended. Our very own Rick Chin’s keynote session on “The Unified Communications Shootout” had a couple hundred people in the room (cool format…attendees hear a six minute pitch from four or five vendors touting why their approach in the best. The attendee can then decide from that whose booth their going to go spend time at getting more info).
Moshe Maier - The Flat Planet - Is VoIP Recession Proof?
Well based on the number of visitors to our booth last night at the ITEXPO opening night, the answer is yes!
Considering Convergence - Network World by Matthew Nickasch:
The session halls and exhibit floor offered standing room only during opening day at IT Expo East 2009, in Miami Beach, FL. With many innovative products, solutions, and services being discussed and displayed, it is clear that all forms of IP telephony and convergence communications are alive and well.
Andy Abramson - Industry blogger - First Impressions of ITEXPO East 2009
VoIP is far from dead and judging by the crowds and the rooms that were full of people the rumors of its death are far from over. I’ll be the first to admit that trade shows in general are down, but this year’s IT EXPO East in Miami Beach Florida is anything but. Walking the floor last evening I counted over 150 booths, and each one was buzzing with activity. In the hallways people were talking and the symposium/conference tracks had rooms full of people.
Garrett Smith - Industry blogger - IT Expo Proves VoIP Industry is Still Going Strong
Well, as usual Rich and crew, did a fantastic job. Yes, the number of exhibitors was down a bit from shows of the past, but the sessions and keynotes have never been as robust.
Not to mention the surprising level of foot traffic that was present from show open to close.
The VoIP industry is not dying. It’s alive, growing and those involved with it continue to impress me with what they are achieving.
In speaking with over 100 of the most influential people in the VoIP industry, the economy has had an effect on their businesses - in most cases a positive one. Someone even told me their customer base tripled from November to January - oh and they’ve been in business for a few years.
Marc Robins - industry analyst IP Communications Insights - Ingate SIP Trunking Seminar is Packed
Apparently bucking the current trend of “downsized” conference attendance, Ingate’s workshop on the first day of the Internet Telephony Expo in Miami Beach is packed with an overflow crowd.
While this is a clear signal that SIP trunking is still a red hot IP communications topic, it is also a very promising indicator of heathy attendance at the show in general.
Bill Miller, Digium: Facebook wall post
Awesome event! Digium Asterisk World Event is absolutely high quality attendees, presenters, and we thank everyone participating!
Tony Rybczynski, Nortel - Dispatch from ITEXPO
Earlier today, I presented in an ITExpo keynote session on UC. Surprisingly good attendance and some probing questions on mobility, converged networking , CEBP and the future of the desktop phone.
I was very happy to see that IT Expo seems to have fought off the economic slump. Maybe that’s a good omen.
O yes, it’s been running for 10 years, so congrats to Rich, Greg and the gang. I think I spoke at the first one and many in between.
As Greg stated, if ITEXPO East 2009 can be considered a barometer of how the VoIP industry is doing, then it’s certainly not a stretch to say that the industry is far from dead. 2009 won’t be the year VoIP died, but the year VoIP thrived!
Tags: andy abramson, bill miller, conference, garrett smith, greg galitzine, itexpo, joe staples, marc robins, matthew nickasch, moshe maier, Tony Rybczynski, tradeshow, voip
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