Logitech today unveiled its first Mac-compatible webcam with premium autofocus technology and Carl Zeiss optics - the Logitech QuickCam Vision Pro webcam for Mac. The new Logitech webcam of course works with iChat and Skype and features a 2-megapixel camera sensor.

The new Logitech webcam uses a voice coil motor for its autofocus system, instead of a stepper motor. According to the news, "Focusing is fast and fluid - crisp even in extreme close-ups only 10 cm from the camera lens. Logitech's autofocus system compensates for changes in image-edge sharpness and refocuses images in less than three seconds"

What I don't get is why Logitech chooses to make operating system specfic webcams. Isn't USB supposed to be an industry standard? I should be able to take any USB device and plug it into a Mac, Windows XP, or Vista, and it should have drivers available that just work. I have plenty of headsets and USB VoIP devices that work on multiple operating systems. I just don't get the point of operating system specific webcams.

In any event, similar to the Logitech Quickcam Orbit AF I reviewed, the new Logitech QuickCam Vision Pro webcam for Mac leverages Logitech's RightLight 2 Technology, which enables the webcam to adjust intelligently in dim or harshly backlighted situations.

Features:

• VGA-quality video at up to 30 frames per second
• the QuickCam Vision Pro records sharp video clips in the 720p high-definition (HD) video format (960-by-720 pixels).
• Complementing the style of your Mac computer, the new webcam for Mac computers presents a black-and-silver profile with a prominent built-in silver microphone.

Pricing and Availability
The Logitech QuickCam Vision Pro webcam for Mac is expected to be available in the U.S. and Europe beginning in July. The suggested retail price to buy one in the U.S. is $129.99.

Asterisk integration with Office Communications Server 2007 is a popular topic in the unified communications community. You wouldn't think integrating an open source Linux-based solution (Asterisk) with Microsoft's proprietary OCS 2007 UC platform would be popular, but it is. In fact, I have my ear to both the Microsoft OCS 2007 community and the Asterisk community and the integration of these two worlds is something I hear all the time. Some within the Asterisk community poo-poo the thought of integrating with Microsoft OCS 2007. Now, whether it's because of some anti-Microsoft zeal or that Asterisk "can do anything - just give the open source community time to build it" mentality, I have to respond that OCS 2007 is here to stay and is doing very well. I am aware of several Fortune 1000 companies, including security-conscious major banks, that have deployed OCS 2007.

Fortunately, not all within the Asterisk community have such blind hatred for Microsoft that they're willing to try and integrate with Microsoft OCS 2007, arguably the best UC platform on the market today. There are some features that Asterisk just doesn't have, such as a feature rich softphone client with video support. The Office Communicator client integrates presence, call control, IM, collaboration, video, and more. Here's a sample of me using it.



Of course, you can use X-Lite, a free SIP client, with Asterisk, but it's functionality and integration isn't as feature-rich as the whole OCS platform. Though if price is your only concern, Microsoft's OCS licensing may be too much for some.

So in any event, what's the difficulty in integrating Asterisk with OCS 2007? Well, as I have written previously, the main issue is that Asterisk uses SIP over UDP and OCS 2007 requires SIP over TCP.

There are some workarounds however. One is to install an unsupported patch onto Asterisk that adds SIP over TCP support. Personally, I'd avoid this though, since any future updates could negate the patch and potentially break the "official" Asterisk. Further, SIP over TCP isn't very common (except in OCS 2007 environments). A better alternative is to install sipX, an open-source SIP proxy which handles both SIP/TCP and SIP/UDP and acts as a gateway between Asterisk and OCS 2007. You can use OpenSER as a SIP UDP to SIP TCP gateway as well. Basically it looks like this:
Asterisk < SIP/UDP > OpenSER < SIP/TCP > OCS Mediation Server < uaCSTA:TLS|TCP > OCS Server

On a related note, pbxnsip, a Linux-based IP-PBX platform offers nice & easy integration with OCS 2007. Check out the pbxnsip with OCS 2007 wiki. You can also use the low-cost pbxnsip appliance as an inexpensive SIP UDP to SIP TCP gateway if you want and use this in conjunction with Asterisk. In fact, pbxnsip can be easily integrated OCS 2007 and with Exchange 2007 UM (Unified Messaging). Check out my full review of pbxnsip for some background on the product.

A new project http://sourceforge.net/projects/asterisk-dotnet (Asterisk LCS gateway) to OCS 2007 claims to offer Asterisk to OCS integration, but it seems like a work in progress.

This is strictly from the rumor-mill, but I heard that SIP over UDP support "may" be coming in the next release of OCS 2007 - rumored to be called OCS 2007 R2. That would negate the need for a TCP-to-UDP SIP gateway and allow for direct Asterisk to OCS integration. Further, Asterisk 1.6 (still beta) has SIP TCP built in - and some have successfully gotten it to work directly with OCS Mediation server. So even if Microsoft doesn't offer SIP over UDP support in the next release, it looks like Asterisk is adding SIP over TCP support.

I should mention that Asterisk lacks CSTA support and unfortunately OpenSER doesn't support it either. While CSTA isn't required for OCS 2007 to Asterisk integration, CSTA provides tighter integration and advanced features such as RCC (Remote Call Control), presence integration, and dual-forking with presence gateways. Further, CSTA provides for 3rd party CTI applications, including predictive dialers, advanced ACD, and IVRs that leverage CSTA for communicating with the PBX.

One solution I discovered that adds rudimentary CSTA capabilities to Asterisk is offered by M-Networks. Their solution, called the Unified Call Control Gateway, functions as an intermediary that allows the Microsoft Live Communications Server (LCS 2005) and Asterisk PBX to communicate. In addition to remote call control, the software provides for PC to phone calling that allows users to use their computer as a phone. While it was designed for LCS 2005, in theory the software could work on its successor, OCS 2007. Here's some of its features:

• TCP <-> UDP SIP Proxy - Allows PC to Phone calls from Communicator 2005 without the need for additional software (ie. OpenSER)
• SIP URI normalization - Converts sip: or tel: URI's to Asterisk extensions. Also allows for proper caller id presentation for PC to internal extension calls.
• e.164 support - Full support for the e.164 numbering standard

Though, if I understand this product correctly it runs silently in the background as a Windows service processing CSTA requests from Office Communicator to control your Asterisk PBX. Thus, EACH desktop would have to install this software, which is a deal-breaker for me. I'd rather have a server-based software or hardware solution.

It's important to point out that the CSTA standard provides for 3rd party CTI applications including advanced ACDs and IVRs which uses CSTA for communicating with a PBX. This is critical in large call centers. Asterisk not providing CSTA functionality certainly isn't the death knell for providing integration with other application. Asterisk supports their AGI/Fast AGI (Asterisk Gateway Interface) which allows developers to write applications.

Further, the Asterisk Manager API adds third-party call control and can be thought of as a CSTA/TSAPI equivalent. Some Asterisk-based solutions offer TAPI-based desktop call control as well, but TAPI won't help you with OCS 2007 integration - specifically with the OCS Communicator presence/VoIP client (shown to the right).

Let's look at the routing once more:
Asterisk < SIP/UDP > OpenSER < SIP/TCP > OCS Mediation Server < uaCSTA:TLS|TCP > OCS Server

I think rather than this convoluted approach developing a CSTA-to-AMI (Asterisk Manager Interface) could offer better integration with Asterisk. Since OpenSER is not dialog stateful, you cannot send the status of the SIP clients to OCS needed for presence and other functionality. It cannot for instance know when a particular session has been terminated. Asterisk on the other hand is dialog stateful and is aware of all sessions at all times. With uaCSTA interaction between OCS 2007 and Asterisk the user-state and the VoIP-phone-state can be exchanged.The big question is will anyone in the Asterisk community write the code to do CSTA-to-AMI? Anyone?

Of course, if you must have CSTA for the advanced telephony features today, you could look into a SIP/CSTA gateway. There are several on the market. UniGone TelServer offers one. Mitel built one specifically with OCS 2007 in mind but I believe it works specifically with their IP-PBX - same goes for Nortel. I believe SIP/CSTA solutions that are "IP-PBX agnostic" are also available from Audiocodes, Dialogic, and Quintum.

I should point out that OCS actually uses uaCSTA (user agent CSTA) which refers to transporting ECMA-323 (CSTA XML) messages over a SIP session. uaCSTA leverages SIP mechanisms to provide an extensible set of features to support applications. uaCSTA can be implemented by several different types of SIP user agents, including directly by a SIP user agent on a SIP phone. uaCSTA can also be implemented by a SIP B2BUA to augment third-party call control functionality or by a proxy server that is front-ending a PBX.

I also happened across a Microsoft TechNet thread that said, "I work at a company that's developed a SIP/CSTA gateway for Asterisk and MS OCS. It's a standards compliant TR/87 gateway that uses the Manager API in Asterisk and SIP to MS OCS. It can be used for remote call control (CTI integration), location based forwarding and device monitoring." The poster is from LiteScape.com, a company that Rich Tehrani wrote about as well as other TMCnet writers. So this is another potential SIP/CSTA solution. I sent an email to LiteScape for more info but am waiting for their reply.

Conclusion:
Asterisk integration with OCS 2007 is happening today, but you have to jump through hoops to get it to work. Further, if you want advanced features like RCC and dual-forking, you'll need a SIP/uaCSTA gateway -- some solutions of which I previously mentioned. SIP over UDP support "may" be coming in the next release of OCS 2007 which would negate the need for a TCP-to-UDP SIP gateway and allow for direct Asterisk to OCS integration. Also, Asterisk 1.6 beta has SIP TCP built in and some have successfully gotten it to work directly with OCS Mediation server. You'll still need a SIP/uaCSTA gateway of some sort unless some entrepreneurial Asterisk guru writes a CSTA-to-AMI add-on for Asterisk.

Lastly, I'm told that pbxnsip will have uaCSTA support very shortly. Although pbxnsip isn't Asterisk, it is another open source IP-PBX solution and a very feature-rich one at that. It's very easy to integrate with OCS 2007 simply by adding a SIP trunk. The pbxnsip wiki explains the steps very well.

[Correction: I mentioned pbxnsip was open source. This was due to a conversation I had with pbxnsip about them 'potentially' making their solution open source. They are currently not open source, however, I should point out that it is developed in C++ and is therefore easily compiled to run on multiple operating systems, including Windows, Linux, etc.]

Here's a screenshot of the main step performed via the pbxnsip web admin:

Nerd Vittles had been lamenting the fact that the "Asterisk 1.6 development seemed to be on a collision course with the dinosaurs because of developer insistence on deprecating and removing commands from the application programming interface (API) in the name of technology enhancement."

Nerd Vittles then explains why this is a bad idea.

The problem this poses is that applications and dialplans written for previous versions of Asterisk no longer function even though the code is barely a year old. In the corporate and government sectors, this would mean major, costly (annual) rewrites of code just to keep a functioning phone system. And, as we noted, these organizations buy phone systems to last a decade so such a development strategy would all but rule out use of Asterisk in the Fortune 500, medical, and government sectors.

True that!

Nerd goes on to sound the alarm regarding the future of Asterisk.

Today we want to share the Digium response and address some of the new issues that have been raised. For those of you that haven't met him, Jared Smith, who co-authored the terrific Asterisk: The Future of Telephony books, now serves as Digium's Community Relations Manager. Jared sent us a thought-provoking response which you can read in its entirety here. For ease of understanding, we're going to quote a number of sections of Jared's response and address them below so that you get the full picture of how dangerous the Digium development approach is to the future of the Asterisk project.

Go read Nerd Vittles entire post.

The main premise of Nerd Vittles post is that it would seem that Asterisk, the "Incredible Hulk of Open Source Telephony" is "breaking" the cardinal rule when it comes to programming - namely don't break your existing applications and be backwards compatible.

But apparently, the programmers at Digium have been breaking existing applications. I contacted Digium creators of the "Incredible Hulk of Open Source Telephony" for comment... Their spokesperson The Hulk was nice enough to comment.

Me: Hulk, I'd just like to say that I love open source Asterisk, which was created in the good ole' USA!



Hulk: Hulk like USA except when USA Army tanks and helicopters shoot Hulk. Hulk like open spaces so Hulk jump high. Hulk like open source too.

Me: What are your thoughts on Nerd Vittles claiming Digium is breaking existing Asterisk applications in the new Asterisk 1.6?

Hulk: Me hate puny Nerds. Nerds weak.

Me: Yes, but what about his claim you are breaking Asterisk applications?

Hulk: Me Hulk. Me like break things.

Me: Yes, but if you break Asterisk, you will no longer be the "Incredible Hulk of Open Source Telephony"

Hulk: You threaten Hulk. You make Hulk very angry.

Me: Whoah. Calm down there big guy. I'm just sayin'

Hulk: You no say! You not make Hulk angry.

Me: Digium's Jared Smith said the following, "APIs change when major versions of the software are released. (APIs are Application Programming Interfaces -- think of them as building blocks inside of the Asterisk code that both Asterisk and third-party programs can use to do various things.) The problem is, when we make Asterisk better, we often have to change those APIs to do so. What are your thoughts?

Hulk: I like building blocks. Hulk like smash blocks of buildings!

Me: So you like breaking the building blocks of programming? That's not a good idea. Do you know that proprietary PBXs often "locked you in" forcing you to use their solutions?

Hulk: Hulk no like being locked in. Thunderbolt Ross try capture Hulk. Make Hulk very angry.

Me: Nerd gave an example of how the new version of Asterisk (1.6) breaks existing dialplans and why that's a bad thing. He writes: We defy anyone to explain why "making Asterisk better" required breaking every dialplan on the planet because some developer thought Set(TIMEOUT(digit)=timeout) was a code improvement over DigitTimeout(timeout).

Hulk: You confuse Hulk. Me no understand.

Me: Exactly. Nor do the programmers that wrote code & APIs one way and now have to rewrite their code.

Hulk: That not good?

Me: No it isn't. Nerd Vittles succinctly explains:

Asterisk developers can't and won't be responsible for making sure they don't break existing applications and dialplan code, and Digium won't do anything to migrate existing code to new platforms. I'm not sure I understand how development of a piece of migration application code requires a knowledge of every third-party application in the universe. Presumably, the Asterisk development team does know when it changes the syntax of some command in the existing API. Why then would it be so difficult to provide another application that translated the "old code" into the "new syntax?" That doesn't require that any third-party apps be reviewed. And it doesn't stymie future development. Just provide the tool to fix stuff that you broke!

  Any comments to this Hulk?

Hulk: Digium broke stuff? Only Hulk break stuff. Hulk mad!

Me: You should be. Only the Hulk should be breaking things. In fact, I know you only want to "break" the competition. You want to smash Cisco, Avaya, Nortel, don't you, Hulk?

Hulk: Yes! Hulk crush Cisco, Avaya, Nortel! HULK SMASH!




Nerd Vittles has some valid points about Asterisk 1.6.0 breaking existing APIs and I myself as a programmer am at a bit of a loss as to why Digium would do that. But to be fair to Digium, you should read Jared Smith's entire response posted on Nerd Vittles' article, which presents Digium's point of view on why they did what they did. What are your thoughts on Digium changing APIs - often a simple syntax change - that breaks existing applications?

PIKA T1/E1 and analog boards are now compatible with FreeSWITCH.

Some good news for PIKA this morning worth reading...

PIKA Technologies Inc., a developer of media-processing hardware and software, today announced that its entire family of analog and digital host media processing (HMP) boards is now compatible with FreeSWITCH, an open-source telephony platform typically used by large telecommunications companies deploying soft switches and other voice applications.

This is the latest partnership that PIKA has announced as part of its mission to support open-source-telephony application developers, having previously announced compatibility with the Asterisk platform.

"PIKA is thrilled to be partners with the FreeSWITCH community. This will mean more options for application development companies across the globe who are building FreeSWITCH-based solutions," said Terry Atwood, vice president of sales, marketing and customer care at PIKA. "Now, FreeSWITCH users will have access to PIKA’s robust media-processing software and hardware, and as a result, be able to stabilize as well as add incremental value to their applications."
FreeSWITCH is designed to facilitate the creation of voice and chat-driven products scaling from a soft phone up to a soft switch. It can be used as a simple switching engine, a PBX, a media gateway or a media server to host IVR applications using simple scripts or XML to control the call flow.

"I was really excited to get PIKA hardware working in FreeSWITCH," said Anthony Minessale, Lead Developer and author of FreeSWITCH. "It’s a welcome edition to our spec sheet. It’s been long anticipated by our user base and I’m pleased to announce we have it working in time for our May 26th 1.0 release. I’m looking forward to meeting with PIKA in person this August at Clue Con 2008 http://www.cluecon.com."

The PIKA product suite comprises a range of hardware and software designed to support applications built on open-source platforms. These building blocks are designed to provide cost-effective and reliable network connectivity to bridge between legacy TDM networks and open source-based IP telephony applications. With the PIKA Daytona analog FXO (trunk) and FXS (station) boards, and the PrimeNet digital T1/E1 boards, FreeSWITCH developers can select the hardware that meets their application’s particular needs.

ISPBX launched a family of Asterisk appliances with CogoBlue Asterisk GUI tools. The CogoBlue GUI enables the configuration of a complete PBX using simple “drag and drop” icons. So simple even grandma can setup an Asterisk system. This drag-and-drop telephony brings back memories of my days testing CTI application generators from Envox, Pronexus, Artisoft, and others in CTI Magazine & Communications Solutions Magazine. Ah, those were the days... Before this whole VoIP thing got started you had PC-PBXs, TAPI, and TSAPI to provide computer telephony integration. Now it's so much simpler. The PBX resides on the IP network so application development and integration is so much simpler.

Anyway, according to ISPBX, "Programming a pbx is now as simple as selecting an object and dragging it into the ‘viewing pane’ in the position you want. This includes trunks, handsets, voicemail boxes, ring groups, conference rooms, auto attendants." They went on further to explain:

What this means is that it’s visually impossible to make a mistake. Installations will always be 100% right first time every time; and that when someone is called in to make a “small change” at a later date – there will be no unintended consequences.

All our sleek, compact hardware utilizes solid state, no-moving-parts technology, making it trouble free and energy efficient. No drives, no motors, no failures. Ever. That’s how we can promise the most reliable phone systems in the industry. And none of our equipment requires buying specialized, overpriced phones. Every ISPBX unit supports nearly every VOIP phone on the market or even your easily retrofi tted analog phones. Our entry-level model accommodates 15 simultaneous calls and 30 extensions. It supports any VOIP phone that utilizes SIP protocol (nearly all of them do) or even your existing analog phones with an inexpensive adapter.

There are several models, including the 500 Series IP-PBX, 800 Series IP-PBX, and the 1000 Series IP-PBX. Here's a picture of the 1000 Series IP-PBX:




Here's some screenshots of this new easy-to-use Asterisk front-end GUI. I have to admit this is one of the prettiest Asterisk front-ends I've seen.
<click any image for larger images>

PhoneFromHere.com today announced a 5 year agreement with Digium. They have been working informally in the past. The goal is to improve and commercialize the embedded Java softphone created by Digium engineers. PhoneFromHere.com creates web call me button widgets for use on websites, blogs, social networking sites, etc. that not only enables visitors to call you but it protects your privacy as well.

Also, PhoneFromHere has a cool iGoogle 'Phone Home' widget currently in beta. Simply add the PhoneFromHere.com widget to your iGoogle homepage and use your laptop, headset and internet connection to place calls.

Interestingly, PhoneFromHere doesn't use SIP - instead they use Nomasystems' IAX2 opensource library. The IAX2 protocol was of course developed by Asterisk guru Mark Spencer from Digium with the goal of being able to traverse firewalls more easily & interconnect multiple Asterisk IP-PBXs in branch offices. But lest I fall into a SIP vs. IAX2 debate let me just share the rest of the news after the jump...

C²Call (C2Call) lets you make web-based P2P VoIP phone calls direct from your browser whether you're using Linux, Windows, or an Apple Mac. The software is a Java widget that is loaded temporary from the C2Call web page into your browser each time you use C2Call. Using the C2Call Java widget I spoke with Martin Feuerhahn, Director of Actai Networks Pte Ltd Taiwan Branch. The voice quality was excellent and the latency wasn't too bad either considering I'm in Connecticut and he's in Taiwan!

In Internet Explorer when you first load the widget you'll see that the application has a digital signature that has been validated by a trusted source:


Actai Networks GmbH, the company mentioned in the above screenshot is the parent company of www.c2call.com. In Firefox the app just ran without showing the digital signature verification message. The client is just 1.1MB in size so it's fairly small.

The beauty of C2Call is that you can send a call invitation to anybody you want to talk to simply by sending an email invite directly from the www.c2call.com applet. The recipient can click on the link and can instantly talk to you. This is a really great viral feature that is especially good for non-technical people. For instance, you can invite your grandma to talk simply by emailing her. Unlike Skype which would require her to install the software, register a username, configure Skype, etc., grandma can just click a link and speak to you almost instantly. Here's a sample email invite being sent out to my boss and industry-leading VoIP pundit, Rich Tehrani.



The widget is SIP-based and importantly it uses the Speex codec, which gives the software its very good voice quality. So you're probably wondering how a Java-based VoIP app fares versus a Flash-based VoIP app, such as the Flashphone app I've written about. Well, for one if you decide to use an Adobe Flash-based solution, you have to use the Nellymoser codec, which is not a free codec. Second, Adobe Flash requires the use of a media server, which Martin claims is not as scalable as their true P2P (peer-to-peer) VoIP solution.

Another feature of note is that C2Call can traverse about 90% of firewalls using standard SIP technology. The official launch isn't for another couple weeks, but if anyone wants an invite give me a shout (Contact link above).

D2 Technologies today released their mobile handset solution powered by Google Android.

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 unified communications systems. Looking at the GUI and the feature-specs, this definitely seems like a pretty cool product that I need to get my hands on and test.

Check out the news today.

LAS VEGAS, CTIA Wireless 2008 (Meeting Room 355) — April 1, 2008 — D2 Technologies, the market leader in embedded software platforms that power IP communications, today announced that its mCUE™ converged communications client for mobile devices and handsets now supports Google’s Android, one of the industry’s first open-source mobile platforms. D2’s mCUE product line, coupled with Google’s Android software development kit, now enables OEMs to rapidly develop handsets powered by Android that offer a premium,
graphic- and media-rich user experience.  The mCUE solution delivers device interoperability with enterprise IP PBXs, Unified Communications
(UC) systems and service provider networks, integrating all session types including voice, IM, SMS and email messaging through a converged presence-based communications user interface (CUI).

D2 is unveiling the integration solution at CTIA Wireless 2008, being held at the Las Vegas (NV) Convention Center April 1-3 (Meeting Room 355).

“This offering will give developers a complete turnkey solution for developing multi-mode mobile communication devices based on Android,” said Doug Makishima, vice president of marketing and sales at D2 Technologies.
“There is considerable market anticipation around Google’s Android platform, and our goal is to make it as fast, easy and affordable as possible for OEMs to deliver converged, unified communications devices to meet that demand.”

D2’s mCUE pairs an innovative, patent pending communications user interface with the company’s vPort MP VoIP software platform to address the needs of OEMs and service providers delivering integrated Fixed Mobile Convergence (FMC) and 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.

Mobile devices with mCUE provide users with advanced presence-based and push-to-x control of cellular and VoIP calls, PBX extension feature activation, IM, email, SMS and other features typically only available on PC-based unified communications soft clients. Its revolutionary user interface, built on top of a multi-identity, multi-session, multi-protocol engine, enables users to simultaneously utilize multiple different communications services such as enterprise IP PBXs and UC systems, and commercial VoIM services like Google Talk, Yahoo!, MSN, AIM, and others.

Developed by the Open Handset Alliance, the Android platform is a Linux-based software stack for mobile devices that includes an operating system, middleware and key applications.

No Asterisk fan can do without some Asterisk paraphernalia such as an Asterisk or Digium T-Shirt, Asterisk book, maybe even an Asterisk coffee mug from ITEXPO where lots of Asterisk happenings take place. Well now you can add an Asterisk USB hub to your collection of all-things Asterisk! This four-port USB hub in the shape of an "asterisk" was created by industrial designer Joel Escalona

He writes:

Some of these designs have been fortunate enough to be manufactured, others were just lucky to be prototypes or models and the less fortunate live within a render. So if at some point you are interested in finding out more about any of my projects or you want to produce, buy or distribute one, do not hesitate to contact me to ask for more details about any of my designs.

So Asterisk fanboys (& girls) give Joel a shout if you want to add this Asterisk hub to your desktop. You'll be the envy of all your fellow co-workers that also love Asterisk.

Sangoma Technologies just threw down the gauntlet today by announcing a lifetime warranty on their Advanced Flexible Telecommunications (AFT) product line. Sangoma used to have a five year warranty, but now they've gone all-out with their lifetime warranty - unheard of in the telecom space. Wow! Sangoma is aggressively moving to retain a leadership position in Asterisk telephony cards. Digium, Rhino, Pika, Aculab, Dialogic, it's your move...

Full release after the jump...