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D2 Technologies today released at CES what they claim is the industry’s first embedded mobile convergence software solution for dual-mode phones.  D2’s mCUE mobile convergence software solution combines a communications user interface with the company’s vPort MP VoIP software platform and is targeted towards OEMs and service providers to help deliver integrated Fixed Mobile Convergence (FMC) and Unified Communications (UC) functionality.

mCUE provides a complete embedded software framework for multi-mode mobile handsets for enterprise and consumer use, such as dual-mode cellular plus Wi-Fi phones. Its completely Java-based user interface framework for Linux can be ported to other GUI platforms.

“mCUE revolutionizes mobile communications by tying together the best aspects of PC-based communications, such as VoIP, instant messaging and presence support, with the roaming benefits of mobile cellular and connection speed and quality of in-building wireless,” said Doug Makishima, vice president of marketing at D2 Technologies. “It is a complete turnkey solution for multi-mode mobile communication devices.”

mCUE is interoperable with enterprise IP-PBXs and UC systems as well as service provider networks. Further, mCUE provides enterprise users with mobile access to full directory services, extension calling, corporate IM, and other features typically only available on IP desk phones or PC-based soft phones.

The user interface is built on top of a multi-identity, multi-session, multi-protocol engine, enables users to simultaneously log into multiple different communications services such as SIP, Google Talk, Yahoo!, MSN, AIM, and others.

The user interface offers presence functionality, push-to-X control over all types of sessions including voice, IM, SMS, and e-mail messaging. It includes an innovative presence-based converged contact list for all services and tools for managing multiple accounts, services and networks.

mCUE utilizes D2’s Internet Service Interface (ISI) layer, an interface that enables multiple IM, VoIP and e-mail services as well as concurrent sessions. It can be customized to meet specific OEM application requirements and can be rebranded by service providers looking for revenue enhancing customer opportunities.

D2’s vPort MP VoIP software engine gives mCUE seamless FMC mobility through IMS-SIP and VCC functionality. D2 technologies espouses their extensive experience in embedded VoIP solutions, IP communications, and Java device platforms, and claims they will provide core communications solutions for Google Android, Linux communications platforms, and other OS platforms used in mobile devices.

In addition to Linux, vPort has been ported to a number of other OSs and RTOSs, and will be available on Windows Mobile in the near future.

On December 26th, Grey VoIP reported a security hole in Asterisk - an Asterisk SIP Channel Driver BYE Message Denial of Service (DoS) vulnerability. The vulnerability could allow a remote user to send a specially crafted BYE message using the 'BYE with Also' transfer method to trigger a NULL pointer error cause the target service to crash. In just a week, Digium, the founders of the open source Asterisk platform, released an update to fix the problem.

Every IP-PBX will have its share of bugs and security holes, I get Cisco advisories all the time, so I wasn't too concerned when I heard about this security vulnerability in Asterisk. But what amazes me is the fast turnaround to fix it - and over the major holidays no less!

Kudos to the Digium team and/or the open source community which were quick to react to this security issue! I'd be curious if it was the open source community that issued the patch or Digium. I'll fire off an email to Mark Spencer and see if he can give some insights.

Update (10:16am): Mark Spencer got back to me in 5 minutes. Here's his response
"The issue was reported as simply a bug on the issue tracker, but Digium's development team recognized this was a security vulnerability and provided the fix, as well as testing each branch of Asterisk and backporting the fix to all versions of Asterisk that were affected (all 1.4 based). Thanks for your interest!"

Even though the open source community helps in the development of Asterisk, this certainly goes to show the importance of Digium and their hired programmers to update Asterisk. Again, kudos to the Digium team in fixing this so quickly.

Digium, today announced the one millionth download of Asterisk in 2007. Digium's PR rep emailed me to say they have now completed its 24th consecutive quarter of growth and profitability this year. When I heard that number - 24 quarters, I couldn't believe it's been that long. That's 6 years! Wow, how time flies!

Digium has had a busy year. In July, Digium acquired Sokol & Associates and in September, Digium acquired Switchvox, a leading provider of IP PBX phone systems powered by Asterisk for SMBs.

A hearty congrats to the Digium team! Looking forward to your 2 million download mark!

Wow, just read that Movable Type has gone 100% open source. This from Anil Dash:

As of today, and forever forward, Movable Type is open source. This means you can freely modify, redistribute, and use Movable Type for any purpose you choose. Just want the details and downloads? Skip to the bottom. But you might like the story of how we got here. As of today, and forever forward, Movable Type is open source. This means you can freely modify, redistribute, and use Movable Type for any purpose you choose.

Just want the details and downloads? Skip to the bottom. But you might like the story of how we got here

.

Guess the competition from WordPress which surpassed Six Apart in blog usage, has helped push Six Apart in the open source direction. Question is it too little too late? Has the Wordpress open source community become too entrenched? Hard to say, but I think open source advocates will give Movable Type a 2nd look. As a Movable Type blogger myself, I think this move will only help Movable Type in the future.

Just last month, PBX in a Flash launched. Well, Nerd Vittles aka 'Santa Claus for Asterisk', has been busy in his workshop building and adding features to PBX in a Flash. Well Nerd Vittles has finally put on his Santa Claus hat today to deliver some Christmas goodies a wee-bit early. (14 days early to be exact.)

Nerd writes:

Text-to-Speech Returns! If you've been following Nerd Vittles for a while, you already know that our favorite applications for any telephony server are text-to-speech apps. The idea behind all of these applications is that you can pick up a phone to find out the same information that you could obtain with a web browser, or a television, or a radio... only faster as in instantaneous. These applications also free you from the home sofa. You can dial in for the information using almost any telephone from anywhere in the world. Well, that was the theory. For those that have endured the last year of kitchen-sink Asterisk implementations, you also know that text-to-speech was the first casualty in the migration from CentOS 4 and Asterisk 1.2 to CentOS 5 and Asterisk 1.4. Well, guess what? We've finally resolved the choppy sound glitch and text-to-speech and Flite are back with PBX in a Flash, and soon we'll have support for other text-to-speech applications as well.

So Gift #1 is the return of TTS for PBX in a Flash, an Asterisk-based distro! Gift #2 is that Bluetooth is back! According to Mr. Vittles, the other casualty in the migration to CentOS 5 and Asterisk 1.4 was Bluetooth support. Gift #3 is the return of some powerful weather reporting by airport code, zip code, and city code. But Nerd isn't done with his Asterisk goody bag. Nosiree Bob! He also threw in AsteriDex (a Rolodex type app) and Yahoo NewsClips for Asterisk. Gift #4 is the launch of a script repository (http://pbxinaflash.org/) for easily downloading/installing applications. It's a happy early Christmas indeed! The only thing Nerd Vittles (Santa Claus) forgot to deliver was the Coca-Cola that all the Asterisk lovers will need to stay awake (caffeine fix) playing with Flash in a PBX.

QueueMetrics is a popular call center monitoring application that works with the Asterisk IP-PBX and featuring over 150 quantitative metrics to see what is happening in your call center(s). Some of its main features include the ability to view a detailed report of call center activity, down to each call on each queue, and run reports by single queues, or by user-created queue groups, both on inbound and outbound traffic. You can also listen to recorded calls and see activity statistics and duration by call stage, with daily, hourly (or shorter), weekly breakdowns. Today, QueueMetrics 1.41 was released adding several new features.

Here's what's new:

- New editors for users, classes, queues, agents, locations, call outcome codes and pause codes. They are paged and allow full-text searching. (Bug#198).
- Original position when joining queue is now tracked and new graphs are available. (#225)
- Wildcards available for queue names: the wildcards act by selecting all defined atomic queues or members of (#48)
- Schedule adherence: tracking how many distinct agents have been on the phone during a time slot in Call Distribution graphs (#56)
- The peak and average queue lenght is displayed in the Call Distribution page (#210)
- On the agent configuration detail page, the set of queues the agent is a known member of is shown (#71)
- It is now possible to do live call listening of outgoing calls as well as incoming ones, by using a different section of the dialplan (#182)
- It is possible to have "invisible" queues, i.e. queues that cannot be selected from the main page but used by the wildcard expansion system. (#234)
- Queues can be "autoconfigured" based on the data available in a MySQL partition.

Here's a sample screenshot of the interface:


Finally, a number of smaller bugs have been fixed in 1.4.3:

- Bug #71: Now showing the queue association on the agent's detail configuration page.
- Bug #55: Sometimes far too many calls would be shown on the agent's page. Now maximum is the latest 20 calls.
- Bug #221: The configuration wizard would sometimes crash with some malformed files.
- Bug #205: The agent detail popup would show an error on IE
- Bug #206: When running in cluster mode, technical errors on a cluster partition would be silently "hidden"
- Bug #218: The realtime.all_subqueues setting in the default RPM was wrong
- Bug #226: The queues selected in for the wallboard mode whould have to be passed in lowercase. Now case insensitive.
- Bug #227: The "custom report" page would sometimes crash complaining a "missing view".
- Bug #230: IAX2 channel names would nbot be stripped correctly.
- Bug #238: The CPH calculation was based on Contacts without including Sales. Now correct.
- Bug #55: No more than 20 calls are now shown on the agent's page.

Anyone want to build a carrier-class network? Show of hands - would you rather use "tried and true" hardware gear from Cisco or Juniper Networks and pay through the nose OR would you rather build your carrier network using commercial off the shelf (COTS) components? Well, just a few short years ago, most of you would raise your hands for the former over the latter. After all, who would trust their carrier network using various COTS components from multiple vendors? Whose neck would you strangle if something went wrong or it didn't integrate correctly?

Today, COTS is quickly gaining steam due to lower cost, faster standard processors, and open source software. Rich sent me a link to his post titled COTS to the Service Provider Rescue which piqued my interest. He goes through some analysis and perspective including talking about CompactPCI and Advanced TCA. He also points out an educational webinar coming up which I think I might check out. Here's a brief synopsis:

Innovative telecommunications service providers, NEP’s (Network Equipment Providers) and ISV’s (Independent Software Providers) are leveraging commercial off the shelf (COTS) technology to help reduce the cost and complexity of delivering their Next Generation Network (NGN) services.

The IBM BladeCenter family is a COTS (Commercial off the Shelf) platform providing greater deployment flexibility - Central Office, Data Center, or Customer Premise; seamless integration into existing network infrastructures with support of a wide range of protocols and interfaces; and an increase in the quality of provisioned services, facilitated by a high-speed architecture at substantially lower CAPEX cost when compared with legacy and proprietary platforms. This common infrastructure enables simple standard processes, shared spare inventory, pre-installation of power, cables, racks, and networking to reduce ongoing OPEX costs.

To learn more be sure to check out the upcoming TMCnet webinar titled Building Carrier Networks Using COTS Technology which takes place December 5, 2007 at 2:00 PM EST/ 11:00 AM PT. Ironic, that Blade Network Technologies chose to be a sponsor, considering I “spanked” them and Nortel (who formerly owned Blade) in this uber-popular blog post picked up by Slashdot, TechDirt, O’Reilly, ZDNet, and a ton of blogs.

The multi-faceted Nokia N810 internet tablet has finally arrived on store shelves. This highly anticipated Linux-based gadget has more gadgety features than you can shake a stick at - including a slide-out QWERTY keyboard, built-in GPS, digital audio/video playback, Bluetooth, camera, MP3 player and even Wi-Fi capability for surfing and VoIP calling. Like the Nokia N800 Internet Tablet, its predecessor which runs Skype (with the latest firmware), the N810 will come pre-bundled with Skype, making it one of the most powerful portable VoIP devices out there.

The Nokia N810 gives you a truly portable Internet experience that's actually useful due to its large 4.13" color wide-screen display and touchpad screen navigation. It sports a 400MHz processor, Mozilla-based Web browser, and up to 10GB of memory (which comes as 2GB of internal memory with an optional 8GB memory card combined).

Battery Life:

• Continuous usage (display on, wireless LAN active): up to 4 hours
• Music playback: up to 10 hours
• Always online time: up to 5 days
• Standby time: up to 14 days

Connectivity:

• WLAN standard: IEEE 802.11b/g
• Bluetooth specification v.2.0 . +EDR (profiles supported: HID, FTP, DUN, GAP, SPP, HSP, SAP and OPP)
• USB high speed for PC connectivity
• 3.5 mm stereo headphone plug

I like how the N810 has the camera in the front, unlike the Nokia N800 where it pops out of the side. It also features an auto-dimming screen so when you use the GPS feature in your car, it dims the screen slightly so you aren't blinded by the screen at night.


The N810's browser has a Flash 9 plugin, and supports AJAX, which I don't believe any other mobile device can do, including the iPhone. The N810 also adds support for Windows Media codecs, which gives you access to more video content on the web. Besides Skype and Mozilla, other applications supported include Rhapsody, Gizmo as well as a few games (Chess, Blocks, Mahjong and Marbles). Gizmo, a Skype competitor apparently even supports video chat on the device. Cool!

Retailers where you can buy the Nokia N810 Internet Tablet include Best Buy Mobile, CompUSA, Micro Center, and Nokia stores in New York and Chicago. The N810 is available for a suggested retail price of $479.

Ward Mundy over at Nerd Vittles informs me that tomorrow Nerd Vittles will officially launch PBX in a Flash, a new Asterisk-based distribution that bundles in the best parts of Asterisk along with some cool third-party applications. PBX in a Flash includes Asterisk 1.4.13, FreePBX 2.3.1, Apache, MySQL, PHP, phpMyAdmin, SendMail, Perl, Flite, and more. Ward sent me a preview of the news that he plans to share tomorrow. Ward hinted in a previous post that PBX in a Flash would be coming in November - well starting on Wednesday you'll be able to download and install it. The rationale behind yet another Asterisk distro? Well, Ward explained his reasoning as follows:

As some of you know, we just haven’t been thrilled with the direction of the trixbox project lately. Without boring everyone with a lot of detail, suffice it to say that it’s just gotten a little too proprietary, too closed, and too commercial for our open source, puritanical tastes. So today, with a bunch of help from some really sharp folks, we embark upon a new open source project that we hope will become the best-of-breed Asterisk-based development platform. Our design goals are simple: a very modular system that meets the needs of Asterisk experimenters as well as those looking for a reliable, scalable, IP-based business telephony solution with all the bells and whistles.

I'll save the Asterisk politics for another day and stick to the news. Like Asterisk@Home and it's successor trixbox, PBX in a Flash is a simple bootable .iso image. The .iso download gives you a CentOS 5 Linux implementation which will install the latest, greatest collection of add-on’s. According to Ward, "Once the install completes, you’ll have a high-performance turnkey Asterisk PBX that’s easy to upgrade with a simple migration path to either managed PBX service or hosted PBX service. You never have to migrate if you don’t want to, and the stand-alone product will always have virtually identical functionality minus the peace of mind that comes with managed or hosted PBX service. In short, the stand-alone product isn’t ever going to be crippleware to entice you to migrate. We’ll have more about the managed and hosted options in coming weeks."

The plan is once they have a stable base platform, you’ll be able to choose from dozens (if not hundreds) of scripts to add all of the Nerd Vittles goodie bag: AsteriDex, Weather Reports, News Feeds, Email by Phone, TeleYapper, Telephone Reminders, Podcasts by Phone, and on and on. There will also be fax support, turnkey phone scripts, hosting providers with free DIDs and minutes to get you started, and loads of new stuff from developers who already are working on compatible add-on’s.

Wow! This Asterisk distro is Fully Loaded! Say, wasn't that a Lindsay Lohan movie?

Ward stated that you can easily add features you want and skip the bloatware. Not sure if that was a dig on traditional Asterisk or trixbox. I know in trixbox you can pick and choose which modules to install via an easy to use GUI. Anyway, he also pointed out that you can add your own feature requests to the growing Wish List on the Nerd Vittles Forum.

For those that don’t have a dedicated Linux machine, they've got a VMware version of PBX in a Flash that you can run on Windows XP, Vista, or as a virtual Linux server using VMware. Nerd Vittles will have the full scoop tomorrow, so keep your eyes peeled!

Psst. Shhh. You didn't hear it from me, but if you can't wait till the launch tomorrow to get your hands on the .iso images, just head on over to http://pbxinaflash.net/.

You can grab the .iso images NOW!

You recall my SIP to Skype gateway breaks Skype's Great Wall of VoIP, right? You don't? Well, let me refresh your memory. I wrote:

As most techies know, Skype uses a proprietary protocol and does not support inbound SIP calls. If you ask Skype CEO, Niklas Zennstrom why Skype chose their own proprietary protocol, (which many reporters have asked him), he always gives the same canned reply - that they chose their own proprietary protocol because SIP doesn't do everything they need, SIP has issues traversing firewalls, our proprietary protocol is more flexible, blah blah blah. Even though there are now NAT traversal solutions for SIP that perhaps didn't exist a couple of years ago, Skype still hasn't moved to SIP and it doesn't look like they will.

Part of the blog post rants how SIP-based softphone users can't communicate with Skype users. Well, today I learned that a new Skype-to-SIP gateway called Skype SIP Gateway (PE) 1.0 from Zhink.com was recently released. This product allows two callers, one on Skype and the other with a SIP address, to communicate with each other. Sweet! By configuring this product and any SIP server (such as OpenSER, Asterisk, etc.) correctly, you can come up with many interesting working scenarios.

The following possible scenarios are listed on their website:

• Forward Skype callers who are your friends to your mobile number. This is very much like personal “skypeout”.
• Reject all Skype callers whom you do not know.
• Allow yourself and family members with own Skype accounts to call into interactive voice response system provided by Sip server. With this, you can do things like access your voice mails or dial out to PSTN lines.
• Using multiple Skype To Sip Gateways, you can now allow Sip users on different private Sip servers to communicate, using the gateways to provide internet connection via Skype network. This is much like local PBXs of branches of a company connected via internet. Note: This will require multiple PCs, each hosting one Personal Edition of this gateway.
• Allow Sip users on softphones like Ekiga, X-Lite and many others to contact Skype callers directly. The Sip users need not have Skype accounts.

Here are some screenshots of the software:







This screenshot above is interesting because I noticed it is using the jackd daemon. The jackd daemon is part of JACK, "a low-latency audio server, written for POSIX conformant operating systems such as GNU/Linux and Apple's OS X. It can connect a number of different applications to an audio device, as well as allowing them to share audio between themselves. Its clients can run in their own processes (ie. as normal applications), or can they can run within the JACK server."

Obviously, JACK must be that heart of how this application works. It must be muxing the audio from the Skype application into a standard SIP call using the RTP stream. Ironic, that since this requires JACK, this Skype to SIP gateway only runs on Linux - which is often the last operating system to get Skype updates. The software allows you to talk for 1 minute for free without registration. It costs $25 to register. A cool application you should try out.