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Digium today released AsteriskNOW, a turnkey easy-to-use version of Asterisk with a web-based GUI that they claim can get a working version of Asterisk up-and-running in 30 minutes. AsteriskNOW is Digium's answer to the popular trixbox "plug and play" Asterisk distro. Previously, AsteriskNOW was in beta, so today is the first day that it is available as a general release. The AsteriskNOW GUI lets users add, modify and delete users. Download options include ISO/CD Image, VM Player image, Xen universal guest image and LiveCD (burn and boot).


I had a chance to talk with Bill Miller, the VP of Product Management of Digium earlier today to talk about this news. Bill Miller told me, "The most important feature of the AsteriskNOW is the setup wizard, which simply takes the user through setting up the time and date band calling plans, connect to the service provider and off they go." He added, "One of the things you will see that we're in the process of doing is adding a series of service providers that are partners of ours and add it to the GUI. The user can select their own service provider if they already have a VoIP service provider or they can get one directly from our GUI and it will be preconfigured. You click on it and you have 'One Click VoIP services'.

Bill mentioned they are working closely with Polycom so that their IP phones can be auto-provisioned. In addition, he mentioned they will soon be adding an affiliate link "Buy Now" on the GUI so that users can click a link and easily buy a preconfigured Polycom phone to work with Asterisk. It will come with the firmware download to work with Asterisk. With some ITSPs you'll see a combination firmware load that comes with the ITSP and Polycom pre-configured. I asked if a premium was added to the phones if you buy a preconfigured Polycom phone and Bill said "Of course any Polycom phone will work because they're a certified partner. It's actually going to be up to the channel not us if they want to charge a premium for that provisioned phone." Fonality, a competing Asterisk solution charges $50 for their pre-configured phones.



You can do a single click upgrade from AsteriskNOW to Asterisk Business Edition which comes with support, the latest bug fixes/updates, etc. Bill mentioned, "Today everything in Asterisk is pure GPL, but we will be adding partners. Our intent is to add business class partners, we're not going to put everything but the kitchen sink in there. It is going to be very well selected and all built around productivity and mobility over time."

"Digium appreciates the initial response to AsteriskNOW. We placed the initial beta version on AsteriskNOW.org last month and have experienced over 2,000 downloads per day without any promotion! Our goal of reducing the complexity of installing and using Asterisk will expand the market for Digium into more mainstream use," said Mark Spencer, president of Digium and creator of Asterisk. "Unlike other Linux distributions used to deploy Asterisk, AsteriskNOW does not have unnecessary components that could compromise security or performance."



AsteriskNOW, based on the recently released Asterisk 1.4, is Digium's open source software appliance and is available for download on the official AsteriskNOW website

I recently met with BlueWave Telecom, a company just coming out of "stealth marketing" to learn about VoIPFlow 2.0, a software platform that enables service providers to provision and manage hundreds of virtual PBXs leveraging a 64-bit version of Asterisk. Basically, VoIPFlow 2.0 enables 100% hosted Asterisk IP-PBXs. It actually runs virtual instances of Asterisk for each customer using open-source virtualization software. Since each Asterisk instance is "virtualized" if one instance crashes or fails, it doesn't bring down all the other Asterisk instances. VoiceFlow also has clustering for redundancy.

One of their key advantages is that they've developed an interface to manage hundreds of virtual Asterisk PBXs making it easy for service providers to manage their customers. Their user interface is abstracted from the underlying Asterisk config files and allows the user to completely provision and manage the PBX via an intuitive user interface. Customers can be given limited access to various configuration settings as well.

It features what they have trademarked as "One Click PBX Deployment" (Amazon lawsuit anyone?), which grants the ability to quickly deploy a fully-functional Asterisk PBX. They claim to have advanced the concept of hardware nodes as computer resources, and claim VoIPFlow has the most advanced architecture on the market for managing many virtual PBXs at once.  System wide settings are available to manage your entire PBX inventory. According to BlueWave Telecom, "VoIPFlow runs on commodity Intel or AMD servers and you may host the solution yourself in your own data center, or you may choose to have us host it as an on-demand solution." They use Level3 for SIP-based VoIP origination and termination, so essentially with this hosted solution you are also using SIP trunking for the voice. And since they're using Level3, they are fully e911-enabled.

How does it work?
You're probably wondering "if this is a 100% hosted Asterisk solution what sits at the customer premise?" Well, technically it's 99% hosted, since you do have to have their Edge 801 integrated T1 router sit at the customer premise. The Edge 801 is an embedded Linux device that actually runs a paired down version of Asterisk so that extension-to-extension calls are local and can continue to work even if the data T1 goes down. Technically, you don't have to use their Edge 801 device - you can use a Cisco T1 router or what have you, but you lose the additional functionality and redundancy. The Edge 801 is only $1200, so it's probably worth the extra redundancy for local extension calling. It features auto-provisioning of Cisco, Sipura, and Polycom phones - all without mucking with the phone's firmware. In addition, all calls are encrypted and they feature full NAT traversal support and G.729 & G.711 codecs. VoIPFlow can scale to <100 phones since running on a single T1 data line.

Although plenty of users have downloaded the 64-bit version of Asterisk, BlueWave Telecom claims they are the only commercial company running on a 64-bit version of Asterisk. BlueWave Telecom explains their architecture as follows:

"VoIPFlow is built on a custom 64bit Linux kernel called the BlueWave kernel. We hand build our Linux kernel so that we know exactly what is inside - nothing extra - just the core of what we need to provide a screaming fast OS. Next, we utilize a custom virtualization layer that allows every PBX that we spin up to believe it is running on its own server, complete with a Linux process.  The overhead for our custom virtualization layer is less than 2%...very efficient. On top of that we layer our security and licensing modules. These modules protect our intellectual property and provide us a secure and undeniable way to do accurate billing, inventory, and software updates to our licensed systems. Then we layer on a pluggable PBX module. We have built the system so that it may take advantage of any open PBX system on the market that supports open standards. In this release we have chosen to include Asterisk in our distribution. Next, we hand build the custom logic necessary to provide fail-over, load-balancing, and auto-leveling. Finally we layer in a beautiful, CSS-based user interface to tie it all together. Our architecture is built for scalability. In order to scale, you simply plug in another hardware node and the system can be rebalanced. You can run several hundred of our virtual PBXs on a single hardware node."

BlueWave Telecom is one Asterisk solutions provider to keep your eye on in 2007. Though this reminds me of my "Signate, an Asterisk provider, bites the dust" article where Signate, an Asterisk provider went bankrupt. I said in this article "Is this a case of the Asterisk ecosystem growing too fast, with too many players trying to get into the Asterisk game? With free Asterisk offerings such as AsteriskNOW, trixbox, and inexpensive solutions from Fonality, Digium, and other providers - some hosted - some CPE, it will be interesting to see who survives and who doesn't. Considering Asterisk is an open-source solution, adding enough value and margin to Asterisk is a tricky business to remain profitable, especially when you consider that open-source advocates tend be very thrifty (cheap?) when it comes to paying for software."

Is BlueWave Telecom going to be a major player in the Asterisk market or can we expect other Signates within the Asterisk ecosystem? I won't make a prediction, but will say I like that they're running Asterisk on a 64-bit Linux kernel with clustering and redundancy. My reservation isn't about them using Asterisk or their technology in general - I just haven't seen the hosted IP-PBX market take a sizable market share. Is the hosted IP-PBX market ready to explode in 2007? If so, BlueWave Telecom's technology could put it at the forefront.

speaQ is a new softphone application designed to make VoIP calls using WiFi or EVDO on mobile devices. Created by QTech, Inc., they claim it was designed from the start for smartphones and PDAs, which have limited processing power. It currently runs on Windows Mobile 5.0 Devices and under Linux on the Sharp Zaurus.

If you have a SIP-based Broadvoice or other SIP-based VoIP phone service account, Alpha Trial speaQ provides a simple phone interface with full call logging, contact integration, and DTMF (touch tones), on any 300Mhz+ Windows Mobile devices, such as the Palm Treo 700w, HP Ipaq 2495, etc. or under Linux on the Sharp Zaurus 5600. There are no time limits on the alpha.

Features include:
- Standard Dialing
- Incoming, Outgoing, Missed call log
- Caller ID
- Last Dialed Number recall
- Mute
- Ring Tones
- STUN support for firewall communication
- Open Standard Telephone Client. (RTP, RFC 3550)
- Session Initiated Protocol (SIP, RFC 3261) signaling.
- Adaptive packet-buffering solution for latency and QoS management.
- G711 codec support.
- Integration with platform address book (Windows, Qtopia) for management of Caller and Called Information.

trixbox 2.0 beta will be available for download on Wednesday. This release will be Fonality's first big contribution to the trixbox/Asterisk community after the recent Fonality acquisition of trixbox. which certainly caused a stir within the Asterisk community. I spoke with Chris Lyman, CEO of Fonality, to find out more about this major new release of trixbox.

First, I should point out that while previous version of trixbox have always been the easiest way to get Asterisk up and running in just minutes, trixbox 2.0 is much more than that.  First and foremost, trixbox 2.0 includes a new 'overall' web GUI to make the whole process "point and click". From this new web GUI you can simply select the modules you want (HUDLite, FreePBX, PHP, lame, etc.) and the web interface will automatically install them. Some of the packages are directly related to Asterisk such as HUDlite or FreePBX, while other options are ancillary, such as SugarCRM. The idea is you shouldn't have to know anything about the command line interface (CLI). In addition, many users wishing to install trixbox want to keep the server as unbloated as possible and not add any unnecessary modules/packages.

Pingtel, a provider of open source, Linux-based enterprise VoIP solutions, and Voxbone, a provider of international VoIP origination services will announce on Monday the completion of interoperability testing between their respective offerings. As a result of this certification, customers can select Pingtel's SIPxchange IP-PBX VoIP software solution in combination with Voxbone's call origination services and thereby benefit from cost-effective calls routed to a SIP-based device (IP-phone, IP-PBX, etc.).

When a customer is in need of an international presence the customer interconnects via VoIP to the closest Voxbone POP. Voxbone then allocates the desired amount of numbers and capacity to the customer. When someone calls to one of these numbers Voxbone forwards the call via VoIP to the customer. Unique to the Voxbone offering is that they only charge a fixed monthly fee for this service - there are no per-minute fees. Here's a diagram explaining the architecture. The POPs are on the left and the SIP device (such as Pingtel) is on the right. The middle is the network/Internet.



Voxbone leases international VoIP virtual phone numbers and worldwide origination services via VoIP to organizations in North and South America, Europe and Asia/Pacific regions. Using either direct inbound dial (DID) or virtual numbers from Voxbone, customers may receive inexpensive, locally dialed phone calls from 50 countries and 4,000 cities throughout the world.

I just arrived in sunny San Diego after a very turbulent JetBlue flight from JFK airport. Although it was a bumpy ride, the 37 channels of DirecTV certainly helped pass the 5hrs and 40 minutes away. After arriving at my hotel, I learned my room wasn't ready, so decided to head on over to the convention center to check out what was happening on the first day of Internet Telephony Conference & Expo. Since I couldn't check in and change into business attire (a suit), I headed over wearing a casual short-sleeve polo shirt, jeans, and sneakers. There should be a rule that VoIP bloggers don't have to wear suits anyway. Not to mention, I'm supposed to be a techie geek that heads up TMC Labs and tests this VoIP stuff!  Maybe I'll ask my boss about that. Then again, he may suggest I wear a white lab coat like he once suggested many moons ago. Not sure which is worse - wearing a suit & tie or wearing a bright white lab coat.

In any event, I arrived at the convention center, headed over to registration, opened my laptop to check email and not 30 seconds later someone recognizes me. Someone behind me says "Tom Keating?". I turn around and not recognizing him say "Yes?" I quickly scan his name badge and see it's Dan York, whose blog I often check out, but whom I have never met. "I take it you recognize me from my blog photo eh?" I ask Dan. Sure enough, that's how he recognized me and which I must admit I was humbly flattered. This isn't the first time either. I've had many people recognize me simply from my blog photo or my photo gallery photos, which always makes it an awkward moment for me as I scan my brain to figure out if I know them or not or if they're regular readers of my blog and have commited my picture to memory.

Dan and I talked some VoIP shop, including Asterisk, trixbox, Fonality, open source, Digium, Pingtel, and a few other items. I meant to ask him if he's going to the bloggers/analysts/press dinner tonight at the Flemings Steak House, being organized by Andy Abramson. Should be fun to finally meet many of the other VoIP bloggers that I often read on a daily basis.

On a related Asterisk note, I wanted to share some news that Switchvox, a provider of IP-PBX phone systems for small- to medium-sized businesses (SMBs), today launched Switchvox v2.6 at Internet Telephony Conference & Expo. Based on the open source Asterisk IP-PBX software, Switchvox has added a lot of feature/functionality to the popular Asterisk solution.

With this release, Switchvox SMB has extended its call monitoring feature and now has even more options, including the ability to jump in on calls or allow colleagues to listen in on conversations. This is a useful feature for call center environments where agents may need coaching, or where a receptionist might need to notify users of a call holding.

The latest version of Switchvox SMB also includes updates to the Switchboard. Switchboard now has more granular control over user and supervisor permissions. Companies can now see who each person is on the phone with and monitor, record or pick up calls with the click of a button. Real-time call queue visibility allows users to see the status of other agents in the queue and statistics, such as how long someone has been on hold.

Other advanced features in Switchvox SMB v2.6 include:

•  Call Recording Backups - Administrators can backup recorded calls and XML details with the same easy FTP procedure that's used to back up Switchvox
•  More Reports - New 3D graphs show even more reports and statistics, allowing managers to better visualize how calls are affecting business
•  Dial by First Name Directory - In addition to Switchvox's dial by last name directory, Switchvox SMB now has the option to dial by first name
•  One Touch Agent Login - Now call queue agents can log in and out with a single key press to start receiving calls
•  Improved Updater - The new updater allows for one-click updates that use significantly smaller update files, so its even faster and easier to get the latest features
•  Complex Network Support - Working with businesses that have remote users connecting over multiple VPNs can pose challenges for IP PBXs. Switchvox SMB's complex network support allows the system to smoothly traverse even the most unusual network scenarios
•  VoIP Provider Compatibility - Switchvox SMB is now even more configurable to work seamlessly with virtually any VoIP provider

Switchvox products are sold as turnkey solutions that include the server hardware and pre-installed Switchvox software. Switchvox SMB is available immediately, starting at $2,495.

The big day is tomorrow and Thursday, but just wanted to get this bit of interesting news to ya. Stay tuned for more...

Fonality will announce tomorrow that they have acquired trixbox, formerly known as Asterisk@Home, and the the world's largest Asterisk-based community. Trixbox is a turn-key, bootable .iso CD image that can turn a PC with no OS into an Asterisk server with a variety of open source tools in just a few minutes. The trixbox application lets someone download a bootable .iso image that then automatically installs Linux, Asterisk, SugarCRM, MySQL, FreePBX, and a whole variety of other applications. Trixbox fully supports the Linux yum command and RPM ecosystem for performing updates and bug fixes.

Essentially, trixbox uses the latest and greatest version of Asterisk. Within 48 hours of a new Asterisk version, engineers being work on the next release of trixbox and they add their own host of patches they they put on top of that. These are basically patches created by innovators, folks inside the Asterisk community that didn't want to sign the rights to Digium via their waiver. The waiver extends Digium's rights to sell the Asterisk code outside of the standard GPL, to which the community must adhere.

I spoke with Chris Lyman, the CEO of Fonality earlier this afternoon to talk about the acquisition of trixbox. He began by wondering how I figured out Fonality and trixbox were working closely together.

Chris: I don't know how you figured out on June 6th that we were getting involved with trixbox, but my hat goes off to you.

Tom: I recalled that I was wondering why Fonality would offer their hudlite, a real-time call control and presence management platform that works with the commercial (paid) Fonality PBXtra. I didn't understand why Fonality would want to make a "free" version of Asterisk (trixbox) more "feature-rich". Curious what the relationship was between Fonality and trixbox, I I did some detective work. Besides googling I also did registrar 'whois' lookups on hudlite.org (Fonality website) and trixbox.org and noticed that the IP addresses were the same - in other words - the same web server. Interesting to say the least, which is what sparked my June 6th post.

In my June 6th post, I pondered, "Did Fonality buy out the rights to Asterisk@Home and then change the name to Trixbox.org? What does this mean for the popular open-source Asterisk@Home distro (now Trixbox) considering Fonality is a for-profit Asterisk solutions provider? Is the plan to try and convert Trixbox users (generally novice Linux users) into paying Fonality users? Fonality certainly focuses on businesses that have little or no Linux experts, so there is certainly a potential synergy there. Well, the mystery continues... I'll post more when I hear back from Chris @Fonality." Chris Lyman and Andrew Gillis responded in a follow-up article.

In any event, it appears the seeds were sown for this acquisition back in June and that I was 4 months early in my pondering whether Fonality had acquired trixbox.

Tom: So what about Digum’s waiver requirement to assign rights of the code back to Digium? Is this an advantage of trixbox since it doesn't have a double licensing strategy?

Chris: There are a number of open source innovators that don't like this dual licensing approach, folks that don’t feel it is in the spirit of being truly open source. These types of innovators make incredible contributions to Asterisk, such as faxing. In fact, the only reason why fax isn't in Asterisk, but it is in trixbox, is because the guy that contributed this code didn't want to sign the waiver and give the rights to Digium to profit from his work. So, basically consider trixbox the latest greatest Asterisk plus a whole lot of innovation.

Tom: So what market are you going after with trixbox?

Chris: So in terms of what market we're going after with trixbox, we're not really going after a market. trixbox is really a community of Asterisk innovators and we're just going to be supporting that community. It's our way of supporting the platform that has been a big part of our success.

Tom: What is the value that trixbox brings to Fonality?

Chris: The value to Fonality is the community value.  The business value that trixbox brings is that there is probably a number of IT directors lurking in the trixbox community that are sort of trying for free, but really do want a commercial company to hold their hand when they roll out. And so we just want to make ourselves known that there is an option to go fully supported on a fully commercial Asterisk based platform over at Fonality..

Tom: So by working within the community you hope to build brand awareness for your commercial-based Fonality PBXtra?

Chris: Yes, we want to build some brand awareness in the Asterisk community to let them know we are a serious player that has a 100% supported, 100% service model.

Chris: The trixbox forums has over 20,000 posts in the last 3 months. It has become the defacto place to get questions answered about Asterisk. Questions answered about rolling an open-source small business environment. And that's really the value we saw is - there are a lot of smart open-source people in that community.

Tom: What are the download numbers?

Chris: 1,500 people download trixbox every day, which is more than Digium. Mark was quoted in a Forbes article as saying 1,000 downloads per day and we were surprised since we averaged 50% more than that.

Tom: Any issues with people knowing about the trixbox brand and knowing that is the latest and greatest version of Asterisk?

Chris: I would say given our download numbers and given the fact that we get more downloads of Asterisk every day more than the rest of the world combined, I would say no, there is no brand problem.

Tom: So how is Fonality going to contribute to trixbox with this investment?

Chris: There's two things that are really really important for us to let the world and the community know. Number one is, trixbox was free, is free, and will always be free. And when I say, I mean pure GPL. It won't have a double license, you won't have to sign a waiver releasing your rights to Fonality, and we're not going to get into any of those complicated licensing schemes that you see with some other open source companies. It will be pure GPL. Number two, we're contributing broad financial support to the trixbox platform to continue to improve that application. This is not just a community of that site that we're going to pay the bandwidth on. We actually have a host of engineers internally working on improving trixbox.

Tom: On a different note, any thoughts about integrating SugarCRM, MySQL, etc. onto the Fonality PBXtra hybrid-CPE-hosted solution for an "all in one box"?

Chris: Now, *that* we *are* very much looking into. We've even had cursory talks with SugarCRM about it. Weve been looking at ways of linking PBXtra and SugarCRM's contact center together on one box. That is a product you will probably see in the future from Fonality. Yes, I will caution any business owner, be careful of how much load you put on a single server since it becomes a single point of failure for your entire business back-office.

Tom: I know the Fonality code is a more secure and stable version of Asterisk but running an older Asterisk codebase, yet without sacrificing functionality. So I was wondering what percentage of code that is in trixbox is going to come back to Fonality?

Chris: Today, it is 0% because our version of Asterisk has been hardened aggressively over the last two and a half years. We think there may be a time if the Trixbox community requests it where we might give our version of our code to the community and call it you know, "stable". But really, more than anything the community wants the latest and greatest features and are willing to sacrifice a little bit of reliability to get there. And so unless we see a great need, we're not going to mix the two different flavors.

end interview...

One final point of note is that trixbox founder Andrew Gillis will join Fonality and continue to lead the trixbox community. The main takeways from this news is that Fonality will commit engineering resources and financial support to trixbox, and just as importantly, trixbox will continue to be 100% GPL without a dual-licensing strategry or a commercial waiver. Trixbox founder Andrew Gillis said, "Fonality shares my vision of making Asterisk free and easy for everyone. They have already proven to me how serious they are by committing a team of engineers to help create the next version of trixbox.

Update:
I had a follow-up conversation with Chris and he wanted to clarify some of his licensing remarks. I made some minor edits to clarify the licensing.

After the news that Zultsys was going out of business, only to hear that they are being resurrected, word from two sources is that a well-known IP-PBX company may be on its last legs. I don't want to disclose who it is at this point without some further investigation. No point causing a company harm from what is just rumor at this point, but I will keep you posted.

This got me thinking though. What happened to the days when there were dozens of PBX manufacturers? Sure there are still many around, but many are hurting, and some have gone belly-up, such as Comdial, Praxon, and others. You have inexpensive open-source IP-PBXs such as as Pingtel and Asterisk that are just as feature-rich as the "big boys" (Nortel, Toshiba, Avaya, Cisco) at 1/8th the cost or less. How can a large company with hundreds of employees and with vastly larger overhead compete with a small nimble company like Digium, the founder of the Asterisk open-source movement?

Will open-source communications systems inevitably kill the major PBX manufacturers? Hard to say, but open-source sure didn't do SCO UNIX any favors when the "free" Linux O/S came on the scene. The days of proprietary communications are over, which also means more competition and smaller margins. In telecom it's SIP that is opening the doors for small start-ups to innovate without being blocked by proprietary and predatory tactics. Only the nimble with the best features, best value, best marketing, and best support will survive the long haul.

On a related note I recently discovered PostPath, a Microsoft Exchange Server alternative, which is the first to implement Exchange network protocols on a Linux email server and the first to let you use your existing Outlook clients with no disruption. According to this article, benefits of selecting the PostPath Server include avoiding vendor lock-in, saving money, increasing performance by 5x, improving resilience, and increasing flexibility and innovation. According to the article, by moving to PostPath you can slash software, storage and infrastructure costs by 75%. We have Exchange Server at TMC and have experienced our share of Exchange Server failures resulting in email loss. Disaster recovery for Exchange Server is just that - a disaster. We've had some outages that took 2 days to entirely fix. Postpath, while not open-source or free, is a Linux-based solution that is less expensive and they claim more reliable with quicker disaster recovery.

Now if only I could have a 100% open-source, IP-PBX, with Exchange Server functionality, built-in web server, Jabber/IM server, collaboration capabilities, mobile phone email synching (e.g. Blackberry), and just about any other communications method, all on a turn-key platform with each component interoperating/integrating - then life would be good.

Paragon Wireless released the world first commercial GSM/VoWLAN dual-mode smart phone, hipi, in March 2006 and was my pick to be a winner in this years' TMC Labs Innovation Awards. With a tri-band GSM/GPRS (Class 10) radio and an IEEE 802.11b WLAN chipset, hipi enables users to enjoy broadband multimedia services at WLAN covered homes, offices, hot spots/zones as well as reliable GSM/GPRS service anytime anywhere. Featuring a 2.4 inch TFT touch screen, QVGA, with 260k Colors, the hipi also supports STUN-based NAT traversal, the SIP standard, as well as G.711, G.729a/b, and G.723 codecs. It even has some gadgety-bling via it's built-in MP3 player and a QVGA/QCIF camera. The hipi can perform SIP-based seamless handover between GSM/VoWLAN. Importantly, it utilizes a unified phone book for both GSM and VoWLAN dialing and a unified GUI for the main applications i.e. phone, E-mail, QQ (IM), and browser.

EXCLUSIVE! The current hipi is based on Linux, but I asked about a Windows Mobile 5 version since there are many popular 3rd party applications written for Windows Mobile and they said that a Windows model was indeed in the works - with some beta trials currently and plans for release later this year. You heard it here first - the first dual-mode Windows Mobile 5 phone out later this year! According to Paragon, the Windows-version will have comparable battery/performance characteristics of their current Linux model. It will be Quad-band phone which means it will work just about anywhere in the world.

The hipi has excellent performance in power management, mobility management, security, mobile VoIP, and voice quality. hipi has passed most of regulation certification programs and has done interoperability testing with over 40 VoIP service providers, system integrators, and infrastructure equipment vendors worldwide. I should point out that Paragon is part of the MobileIGNITE program, an open standards approach to network convergence provides an end-to-end solution that enables communications service providers to rapidly deploy mobile to VoIP roaming services. (Bridgeport Networks is a well-known company that is part of the MobileIGNITE program.)



Standby time >100 Hours (GSM on, WLAN on) >200 Hours (GSM on, WLAN off). Talk time on VoWLAN is 3.3 Hours and for GSM it's 7.8 Hours. With such excellent battery talk times and standby - even with two radios operating - hipi is an ideal device for fixed mobile convergence. Check out the specs...

Hardware Specification (Linux model)
• Intel PXA271 processor with embedded Linux
• 2.4 inch TFT touch screen, QVGA, 260k Colors
• Built-in speaker/microphone, 2.4mm stereo and headset
• 1.3M pixel CMOS camera
• USB slave
• Mini SD
• 1100 mAh Li-ion battery

GSM Specification
• Frequency bands: 900/1800/1900 MHz
• GPRS Class 10
• SMS, MMS, WAP applications
• FTA/CTA certification
• FCC/CE certification

WLAN Specification
• IEEE 802.11b
• RF channels: US: 11, ETSI: 13, Japan: 14
• High-gain internal antenna
• WEP 64/128 bits, WPA, 802.1x
• EAP PSK/LEAP/PEAP/TTLS/SIM
• Power saving modes
• Fast roaming between access points

VoIP Specification
• SIP: IETF RFC 3261
• Codec: G.711, G.729a/b, G.723
• Acoustic echo cancellation
• Dynamic jitter buffer
• Voice activity detection
• Stun-based NAT traversal

Input Methods
• Handwriting Recognition > English > Chinese > Numeric characters
• Soft Keypads > Qwerty > Standard phone dialpad > Symbol

Power Management Features
• Standby time >100 Hours (GSM on, WLAN on) > 200 Hours (GSM on, WLAN off)
• Talk time > VoWLAN: 3.3 Hours > GSM: 7.8 Hours
• MP3 play time > 5.8 Hours (GSM on, WLAN on) > 6.2 Hours (GSM on, WLAN off)

Fixed Mobile Convergence Features
• Simultaneously activated GSM and WLAN air interfaces
• Handling simultaneously GSM and VoWLAN incoming calls
• SIP-based seamless handover between GSM/VoWLAN
• Automatic/manual switch for out-going calls between GSM and VoWLAN
• Automatic/manual switch for data applications using GPRS or WLAN
• Unified phone book for both GSM and VoWLAN.
• Unified GUI for applications (phone, E-mail, browser, QQ)

Call Features
• Call hold
• Call waiting
• Call mute
• Call forward
• Call transfer
• 3-way conference
• Voice mail
• SMS over SIP
• Phone book - (1000 entries with photos)
• Incoming call prompt with picture
• View phonebook during call
• Enter sketch pad during call
• Adjust volume during call
• Auto-answer/flip answer
• Quick silence
• Turbo dial
• Manual/Auto/Earphone redial
• Call history (20 entries) Data Application Features
• POP3 E-mail client (SSL support) > 100 full E-mails with attachments up to 200KB > Document viewer for MS-Office and PDF files
• Web browser: HTML4.01, JAVAScript1.5, SSL3.0, HTTP1.1, CSS1.0
• Instant messaging: QQ Multimedia Features
• Video format: MP4, 3GPP
• Audio format: MP3, WAV, MIDI, AMR
• Picture format: WBMP, BMP, JPEG, GIF
• Camcorder: QVGA, QCIF
• Media Player > Audio: MP3 player > Video: up to 30 frames/second QVGA MP4/3GPP PIM Features
• Calendar
• Schedule management
• Alarm clock
• Voice recorder
• World time
• Currency converter
• Anniversary Other Features
• English <-> Chinese dictionary
• Calculator
• World time
• Notepad
• Sketch pad
• File transfer
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Digium, creator of Asterisk and pioneer of open source telephony, today announced that Mark Spencer, president of Digium, has been named to Inc.com's "30 Under 30: America's Coolest Young Entrepreneurs".

Inc. selected the top 30 entrepreneurs based on their proven ability to run a successful company, manage a company with a novel approach, create a successful or innovative product, and/or otherwise demonstrate their innovative idea in the world of entrepreneurship. Inc.'s article can be found at www.inc.com/30under30.

"I am honored to be included in Inc.'s 30 under 30," said Mark Spencer, president of Digium and creator of Asterisk. "Work has become quite a passion for me and it is very rewarding to receive such recognition."

Congratulations, Mark!