September 2008 Archives

By the end of this year, Nintendo will debut a new version of its DS handheld console that plays music and is equipped with a camera and advanced wireless functions.

This would put the new DS in direct competition with such well-entrenched devices as the Apple iPod and camera-equipped mobile phones.

The price for the new machine is expected to be below 20,000 yen ($189) in Japan, compared with 16,800 yen ($158) for the current model.

Now I wonder what Sony and other folks have up their sleeves in this very, very competitive market segment?

More at Reuters and at ZDNet.

Unlocked iPhone 3Gs are now on sale in Hong Kong direct from the company web site, according to a published report.

Until now, Hong Kong iPhones could only be bought from Hutchison Telecommunications on a two-year contract, the cheapest package being a staggering $188 a month.

Now Apple's local website is offering the 8 gigabyte version of its unlocked phone for $695 and the 16 gigabyte one for $798, letting buyers choose their own GSM service provider. 

The shape of things to come?

Get more here.

For the first time in recorded history, we used our cell phones more for sending text messages than making phone calls! (Knew that day would come ...)

In the fourth quarter of 2007, the average subscriber's volume of text messages shot upward by 64%, while the average number of calls dropped slightly, according to Nielsen Mobile.

Not surprisingly, users with QWERTY-style keypads sent 54% more text messages than those with ordinary keypads. 

Teenagers 13 to 17 are by far the most prolific texters, sending or receiving 1,742 messages a month, according to Nielsen Mobile. By contrast, 18-to-24-year-olds average 790 messages. 

Thanks (as always) to the New York Times for filling my day with news.

Fingers by Anders Bergstrom; check out it here.

German-based Digital Imaging Systems (DIS)  has introduced a camera module for cell phones that for the first time includes 9-megapixel technology.

(Remember when 1 MP was a lot?)

The module delivers image data with a max resolution of 3488 x 2616 pixels through either a parallel interface or optional MIPI interface. Full production is scheduled to begin later in the year.

The unit also contains an array of advanced photographic features, according to the DIS. These include auto-focus, high quality lenses, mechanical shutter and ND-filter with integrated actuators and an easy-to- access platform software, enabling high-end digital still camera quality in mobile phones.

The announcement was made at Photokina, the huge European trade show.

More at Cellular-news. 

Cell phones will replace the personal navigation device (PND) as the primary GPS device by 2011, according to the research firm iSuppli.

The firm predicted that cell phones will then account for 36% of the GPS market, compared with 30% for PNDs, which today account for 50% of global navigational units sold.

The report also predicted that price drops among PNDs will lead to smaller manufacturers selling out to larger hardware companies over the next year or two. (Doesn't this happen in every gadget category?)

Reminds me how cell phones were going to replace digital cameras -- don't believe that has happened yet.  But maybe GPS is a more easily stormed market ...

More at TWICE.

Muxtape.com has changed its tune.

A month after the NYC-based music site was shut down by the Recording Industry Association of America (RIAA) for copyright infringement, founder Justin Ouellette announced that it will re-launch with a new focus.

The six-month-old site had allowed users to create playlists or mixtapes of up to 12 songs and share the lists with friends. According to a message on the Muxtape home page, the site will now become a service for bands to promote their music on the Web.

The revamped Muxtape will join a long list of Web firms like Nabbr, TuneCore and Music Nation, which are trying to cash in on helping aspiring artists disseminate their music online without signing up with a label. It will have a difficult time distinguishing itself from its competitors, industry observers say. 

More at Crain's New York.

Here's how the RIAA views it.

Gizmo5 SIP trunks have always been available in trixbox CE, but it was a manual process. The Gizmo5 team has built a module to be part of the trixbox package manager that allows you to purchase your trunks, see your account balance, purchase more minutes, and automatically setup your inbound and outbound routes. The module is now available via the trixbox package manager and will be built into all upcoming ISO builds.

Additionally, the calling service for trixbox CE is pre-configured to use the Gizmo5 calling network and includes a new UI for easy administration. Also included is a Tech Check system that confirms basic setup of a trixbox CE system and notifies users when new Gizmo modules are available. Finally, the new offering also includes pay-as-you-go and Gizmo5 has also joined Fonality's FACE program (Fonality Authorized Certified Ecosystem) as a Gold partner to ensure its products are optimized and compatible with the trixbox CE platform.

Check out this cool new converged prototype device from Motorola that combines a picture frame with touch-screen, video camera, Bluetooth headset, VoIP, femtocell, and video streaming. A femtocell is a small cellular base station, typically designed for use in residential or small business environments that allows you to use your mobile phone in your home connecting to your femtocell access point.

Femtocells essentially are an alternative way to deliver the benefits of Fixed Mobile Convergence (FMC) without the need for a dual-mode handset. In the Youtube demo video below demoed by Motorola representative Harsha Hegde, you can clearly see they're using the popular Counterpath Xten SIP-based softphone - also shown in the screen grab above. Motorola also demonstrates a femtocell mobile-to-mobile VoIP call, which is pretty cool.

Regional mobile carrier Alltel has launched NuTsie, a service that allows users of almost a dozen of the company's handsets to stream certain titles from their PC's iTunes collection to their cell phones.

This makes the operator the first U.S. carrier to offer such a service, which will cost $4.99 a month, or $19.99 a year.

To stream the protected and unprotected songs in a user's iTunes library, NuTsie does not actually place-shift songs from a user's PC. Instead, the service matches the songs in a user's library to the licensed songs stored in NuTsie servers, then streams only the songs in its server.

As a result, not every song in a user's PC library might get streamed to an Alltel phone or BlackBerry. Partly to compensate for that limitation, NuTsie gives users the option to stream songs not in their iTunes library.

New songs would be chosen by NuTsie based on the user's existing library, whose playlists must be uploaded to NuTsie and will appear inside the phone's NuTsie application. New music is also available for playback from friends' playlists and from NuTsie programmers.

Even more at TWICE.  

Asus has announced that it will be adding 3.75G connectivity to its hugely popular series of Eee PC netbooks, enabling convenient and high-speed access to the Internet anytime, anywhere.

The inclusion of 3.75G is a perfect addition to the Eee PC's existing set of travel-friendly features such as its high portability, shockproof data storage and all-day battery life -- strengthening its reputation as the solution for computing on the go. 

Frequent travelers will particularly welcome the timely addition of 3.75G support, which comes as service providers around the globe are ramping up their adoption of 3.75G High-Speed Uplink Packet Access (HSUPA). This means that they will be assured of a reliable, high-speed mode of Internet access in many destinations around the world.

Read more about it at the Asus web site.

Flashphone is a web-based SIP softphone, while gtalk2voip lets you make or receive calls to/from all SIP phones and SIP services, including Yahoo! Messenger, MSN Messenger, and Google Talk. Both Flashphone and gtalk2voip are free. Now combine the two and you can make free web-based Flash calls to Yahoo Messenger, MSN Messenger, and Google Talk (gtalk) users.

According to the Flashphone blog, "For example, if someone is online in Gtalk and you want to call him from flashphone you just need to enter SIP URI like sip:google_username@gtalk2voip.com and gtalk user will see incoming call. You also can easily call to flashphone from gtalk via gtalk2voip, add contact like [flashphone_login]_at_flashphone.ru@gtalk2voip.com and call to this contact, flashphone will ring if user online."

Pretty sweet!

image of Flashphone during one of my tests:

A new report predicts that ultramobile devices will reach 200 million in unit sales by 2013 and will match the market for notebook PCs.

ABI Research said products such as netbooks and mobile Internet devices, or MIDs, will gross $27 billion by then, with MID units growing in popularity to surpass its rival.

All this from ZDNet.

On the other hand, in a wide-ranging interview, Dell CEO Michael Dell (in photo) voiced suspicion that small-screened netbooks will shake up the computer industry or become the primary computer in developed parts of the world.

"I think it's a second machine in developed countries and a first machine in newly developed countries," he said, according to a report in CNET.

So is it yes for netbooks or no for netbooks?

Time will tell ...

Continuing the coverage of the big Skype for Asterisk news I covered earlier today... In a nutshell, the Asterisk server acts as a Skype-to-SIP gateway, a very popular requested feature, mapping Asterisk SIP-based phones onto the Skype network via the Asterisk Skype channel driver. Technically, you could call Asterisk a Skype-to-IAX gateway as well.

So how does it work?

Well, on an inbound call to your Skype username, both your Skype desktop client rings (if running) and your Asterisk IP phone rings. You can take the call using either your PC's Skype software or your IP phone. Similarly, if someone calls your SkypeIn number, both will ring. Further, if someone dials your corporate auto-attendant, and then enters an extension number, it will still ring both your Skype client and your regular IP phone. That's huge! You can be remote and use Skype as your remote IP phone.

Essentially, Skype becomes a softphone extension of the Asterisk IP-PBX. Although, it's important to note that that outbound calls from the Skype client go through the Skype network and not through Asterisk, so it's not a full-fledged softphone application which does inbound & outbound through the same Asterisk IP-PBX - important for call detail records (CDR) that businesses need.

Also, using Skype for Asterisk you can assign Skype IDs/usernames to an Asterisk call queues. So for instance, you can setup 'tmcsupport' or 'tmcsales' Skype usernames and then anyone in the world can call into these call queues. Skype's rich presence will be integrated into Asterisk, but it isn't currently part of the beta, but should be part of the final release. What that would allow is a remote agent to set their presence to Away or Available and then take inbound calls to the Asterisk queue based on their presence.

[section added since Digium's Steve Sokol explained how to handle transfers from IP phones to Skype usernames.]

We've got a couple of ways to do it. The first and most simple way would be to create a local numeric alias for the Skype name. In that case you simply transfer the call to the numeric alias which then sends the call out the Skype channel. The extensions.conf logic looks like this:

exten => 6101,1,Dial(Skype/ssokol.digium)

In the above example the extension number is 6101 and the Skype name to which the call is forwarded is ssokol.digium.

Another mode of transfer would involve a graphical user interface like the Switchvox Switchboard. In that case the user would simply drag and drop the call on an appearance that maps to the Skype name. Under the covers it would use the Manager API to execute the transfer.

I'm sure that there are a number of other modes or techniques that could be used. Our developer community is very good at inventing clever solutions.
[end section added]

When asked how Skype IP-PBX gateway appliances are affected by this announcement, Stefan Öberg VP & GM Telecom for Skype said, "The appliances that are out there now have built their solutions on standard Linux client. They've used the public API on that and basically are running many instances of Skype Linux client. Obviously, that's not the way the Linux client was meant to be implemented. So those solutions are not scalable or reliable to the extend that businesses would want them to be. The difference with this solution is that we've built it together to scale and to be reliable."

When asked, "What about video integration?" Danny Wyndam responded, "The beta product that is available today does not support video. It is our plan to be able to support everything you can do in Skype through Asterisk. It's just an evolution of the connector to this platform that we can add the video support."

Danny pointed out that in Asterisk you will be able to define calling rules with least cost routing (LCR) and determine if the call should go out through the T1/PRI/analog trunk or over SkypeOut to save on the costs.

When asked, "How long have you been working on this?", Danny answered that they have been in talks for at least 3 years - but very serious for a few months in integrating Asterisk with Skype.

Here's a shot from Astricon showing it in action:

Skype and Digium have hooked up to bring Skype to Asterisk called Skype For Asterisk. Skype For Asterisk launched minutes ago enables Asterisk users to get access to Skype features coupled with the capabilities of Asterisk. For example, the beta version of Skype For Asterisk will allow customers to make, receive and transfer Skype calls from within Asterisk systems using their existing hardware; enable inbound calling solutions like free click-to-call from company websites or virtual offices; and manage Skype calls using Asterisk applications such as call routing, conferencing, phone menus and voicemail.

Hey, I guess I was right in my (Astricon) prognostications earlier today about it having to do with Skype.

The Skype For Asterisk Beta program begins today. Asterisk users, system administrators and developers are invited to apply to participate at http://www.astricon.net/skype

I'm trying to figure out how you transfer a call to a Skype username (i.e. tkeating) using a traditional (Asterisk) IP phone with no keyboard - just a numeric keypad. Of course, maybe the transfer feature is only to other Asterisk extensions or outside phone numbers and you can't initiate calls to Skype usernames. Of course, I'm guessing that you can map inbound Skype calls to usernames to specific Asterisk IP phone extensions.

[section added since Digium's Steve Sokol explained how to handle transfers from IP phones to Skype usernames.]
We've got a couple of ways to do it. The first and most simple way would be to create a local numeric alias for the Skype name. In that case you simply transfer the call to the numeric alias which then sends the call out the Skype channel. The extensions.conf logic looks like this:

exten => 6101,1,Dial(Skype/ssokol.digium)

In the above example the extension number is 6101 and the Skype name to which the call is forwarded is ssokol.digium.

Another mode of transfer would involve a graphical user interface like the Switchvox Switchboard. In that case the user would simply drag and drop the call on an appearance that maps to the Skype name. Under the covers it would use the Manager API to execute the transfer.

I'm sure that there are a number of other modes or techniques that could be used. Our developer community is very good at inventing clever solutions.
[end section added]

Update (1pm): Some other thoughts...
Will Skype for Asterisk work exclusively on Digium's flavors of Asterisk (AsteriskNOW, Switchvox, etc.) or will it also work on trixbox CE, PBX in a Flash, etc? Is the Skype channel driver licensed by Digium or is it a free driver, which can then be used on other Asterisk distros. Since Asterisk offers a free version of their open source solution, I'm going to have to assume the Skype channel driver will also be free.

Update (1:20pm): Some info from TMCnet reporters at Astricon

• Majority of questions were about access to code. Mark says their will be some limited access.
• Caller ID - they say it can work.
• Number portability - Oberg says that is a 'local issue' and not built in to this beta.
• No pricing announced.
• Commercial license model, Not open source.

Update (2:58pm) Additional info from an interview with Skype & Digium:
Continuing the coverage of the big Skype for Asterisk news I covered earlier today... In a nutshell, the Asterisk server acts as a Skype-to-SIP gateway, a very popular requested feature, mapping Asterisk SIP-based phones onto the Skype network via the Asterisk Skype channel driver. Technically, you could call Asterisk a Skype-to-IAX gateway as well.

So how does it work?

Well, on an inbound call to your Skype username, both your Skype desktop client rings (if running) and your Asterisk IP phone rings. You can take the call using either your PC's Skype software or your IP phone. Similarly, if someone calls your SkypeIn number, both will ring. Further, if someone dials your corporate auto-attendant, and then enters an extension number, it will still ring both your Skype client and your regular IP phone. That's huge! You can be remote and use Skype as your remote IP phone.

Essentially, Skype becomes a softphone extension of the Asterisk IP-PBX. Although, it's important to note that that outbound calls from the Skype client go through the Skype network and not through Asterisk, so it's not a full-fledged softphone application which does inbound & outbound through the same Asterisk IP-PBX - important for call detail records (CDR) that businesses need.

Also, using Skype for Asterisk you can assign Skype IDs/usernames to an Asterisk call queues. So for instance, you can setup 'tmcsupport' or 'tmcsales' Skype usernames and then anyone in the world can call into these call queues. Skype's rich presence will be integrated into Asterisk, but it isn't currently part of the beta, but should be part of the final release. What that would allow is a remote agent to set their presence to Away or Available and then take inbound calls to the Asterisk queue based on their presence.

When asked how Skype IP-PBX gateway appliances are affected by this announcement, Stefan Öberg VP & GM Telecom for Skype said, "The appliances that are out there now have built their solutions on standard Linux client. They've used the public API on that and basically are running many instances of Skype Linux client. Obviously, that's not the way the Linux client was meant to be implemented. So those solutions are not scalable or reliable to the extend that businesses would want them to be. The difference with this solution is that we've built it together to scale and to be reliable."

When asked, "What about video integration?" Danny Wyndam responded, "The beta product that is available today does not support video. It is our plan to be able to support everything you can do in Skype through Asterisk. It's just an evolution of the connector to this platform that we can add the video support."

Danny pointed out that in Asterisk you will be able to define calling rules with least cost routing (LCR) and determine if the call should go out through the T1/PRI/analog trunk or over SkypeOut to save on the costs.

When asked, "How long have you been working on this?", Danny answered that they have been in talks for at least 3 years - but very serious for a few months in integrating Asterisk with Skype.

News release after the jump...

Digium announced the immediate availability of the AEX410, a four-port modular analog PCI-Express x1 telephony interface card for use with Asterisk. The AEX410 is a PCI-Express board that compliments Digium's existing PCI-based TDM410 product.

The AEX410 offers analog (FXS) stations and analog trunk (FXO) modules for connecting to the PSTN or analog devices. An optional DSP-based 128ms line echo cancellation for the AEX410 is provided by Digium's VPMADT032 G.168 module. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces.

According to the Digium blog, the naming convention is as follows:
AEX4XYZ

Where X indicates the number of FXS (station) modules (ports)
Where Y indicates the number of FXO (trunk) modules (ports)
Where Z indicates either B for bundles not containing DSP-based echo cancellation or E for bundles that do contain DSP-based echo cancellation.

So for example, here are some sample models, though not limited to just these:
AEX422E <- 2 FXS, 2 FXO, has DSP echo cancellation
AEX440E <- 4 FXS, 0 FXO, has DSP echo cancellation
AEX404E <- 0 FXS, 4 FXO, has DSP echo cancellation

The AEX410 board product utilizes Digium's wctdm24xxp driver file that is part of the Zaptel (soon-to-be DAHDI) driver package. For more info, check out the blog post.