The snom m3 SIP wireless (DECT) phone is one of my favorite VoIP phones. I've been testing and reviewing it for a few months but haven't had time to write up the review until now. First, let me point out that the problem with IP-PBXs is they typically give you a desk phone or a softphone with no real mobility options to walk around, which is critical in some vertical markets, such as retail and manufacturing. Even sales professionals want the flexibility to take calls while roaming the office. In the past, I have used analog telephony adapters to connect my cordless phone to my SIP-based IP-PBX, but the cordless phone lacks multiple lines, call transfer, call conference, call waiting, or even a message waiting indication (MWI). Enter the snom m3, a SIP wireless phone that like a home cordless phone which not only gives you mobility while on the phone, but full IP-PBX functionality as well, including call hold, call transfer, message waiting indicator, and more. In fact, while the caller is holding, music-on-hold is available from the IP-PBX, giving the same business professional experience from a desktop phone.

I should mention that there are WiFi SIP phones, but the battery life on these phones isn't great. snom takes advantage of Digital Enhanced Cordless Telecommunications (DECT), a wireless communication standard which can seamlessly hand off calls as a handset moves between multiple base stations in a large office, but also has superior battery life than WiFi SIP phones. The Lithium Ion battery offers a very good eight hours of talk time and 100 hours of standby. Additionally, DECT devices use the 1.9 GHz band while WiFi uses 2.4Ghz so they don't interfere with one another. DECT also doesn't suffer the microwave oven interference that often plagues WiFi access points.


             snom m3 Main Menu

The snom m3 supports up to 8 different SIP identities (registrations) allowing you to connect to separate IP-PBXs (or SIP service providers) or the same IP-PBX to support multiple lines. The m3 is 2" x 5" and less than an inch thick sporting a nice 1.75" color LCD (128x128 pixels and 65,536 colors), 2.5mm headset jack, and a speakerphone. The headset jack is a nice feature that I haven't seen on any cordless DECT phones. The phone also comes with a belt clip so you can easily use the headset for talking while walking. The m3 is surprisingly very lightweight - much lighter than I would have expected. The phone also has volume controls, the basic 12 dialpad keys, five navigation keys, and two function keys. The snom m3 ships with some documentation, but for real technical details, the snom m3 wiki is the place to go.


The m3 communicates with the base station which is connected directly to your network via a standard Ethernet cable. Once connected and booted up, the base station obtains an IP address from the DHCP server. By default (factory setting), snom m3 phones are configured to use HTTP as the transfer protocol for provisioning, but TFTP can also be used. Since I was testing this with an Asterisk-based trixbox system, I changed the gateway to use TFTP. Also, the snom m3 supports Option 66 on the DHCP server to automatically acquire the IP address of the TFTP server. Nice!

The TFTP boot server address can be an IP address, a fully qualified domain name (FQDN), or an URL. I also created a config file (/tftpboot/m3/settings/0004132A10E4.cfg) on the TFTP server for the snom m3 to download. I was able to get access to the firmware, upload the new firmware to /tftpboot/m3/firmware/ and it automatically downloaded the latest firmware. Even better you can have it set to connect directly with snom's server (http://provisioning.snom.com/m3/firmware/) to download the latest firmware and even set a schedule to automatically grab the latest version.

Features:

• Display: 128 x 128 pixels, 65536 colors, backlit
• Li-Ion battery pack for 20 hours of calls or 100 hours standby
• Range: 50 meters indoors, 100 meters outdoors
• 12 numerical keys, 5 navigation keys, 2 function keys
• Speakerphone on mobile handset
• Polyphonic ringtones
• Automatic registration of handset
• Separate charging cradle for handset
• 8 handsets per base station
• 8 SIP registrations with different servers/registrars
• Up to 3 concurrent calls per base station
• Three-way conference
• Remote setup, password protection
• Open DECT GAP standard

Since the snom m3 supports multiple handsets, this leads to some interesting multi-handset functionality. For instance, the Telephony Settings on the web interface lets you pick which identity (CallerID) each handset will use when making outbound calls. You can also set which handsets will ring on incoming calls for each SIP registration/phone number. Thus, you can have one SIP registration ring your home office m3 handset, another ring your son/daughter's m3 handset, and another phone number be the shared kitchen m3 phone. In fact, the snom m3 supports three concurrent calls per base station so you can receive 3 simultaneous calls to the handsets.


The snom m3 supports the most common VoIP codecs, including G.711u (PCMU), G.711a (PCMA), G.729ab, and iLBC. G.711 is the standard used by traditional phone systems and it features the best voice quality at the expense of more bandwidth used (80kbs), which isn't ideal for some DSL connections that only sport 256kbs upstream. Fortunately, the snom m3 supports G.729a which only use 8kbps at a slight loss of voice quality. iLBC (Internet Low Bitrate Codec), although not as widely supported, is designed for narrow band speech and supports two bit rates, 15Kbps (20ms frame rate) and 13.3 Kbps(30ms frame rate), though the m3 only supports the 20ms frame rate @15Kbps. iLBC yields slightly better voice quality than G.729a yet also has a higher robustness in dealing with packet loss while using roughly the same amount of bandwidth. It also has a more dynamic range of sound than G.729a. So kudos to snom for including iLBC as a choice.



You can also configure various settings from the phone itself, though it's more tedious. The VoIP settings is protected by a PIN / password which defaults to 0000. From the phone you can configure the timezone and it even supports NTP time servers for accurate time. Additionally, you can add contacts, however adding contacts via the phone is a bit tedious. I wished the web interface let me add them there and then it would push the contacts down to the multiple handsets.

So how's the phone's range? snom claims the phone needs to be within 50 meters indoors or 100 meters outdoors from the base station. I walked around TMC's offices and didn't lose a signal. Then I went outside walked about 250 feet and it was crystal clear. Excellent range I have to say. The voice quality of the earpiece was very good and the remote end said I sounded very good during my test calls. I also tested the speakerphone, and although it wasn't the best voice quality, I didn't expect a fantastic sounding speakerphone on such a small handset. I should mention that you can also perform intercom calls to either a single m3 handset or you can intercom page all handsets. Useful if you are trying to reach someone and don't know where they are located.

Ratings Score Installation Documentation Features Usability Performance Overall

All in all, the snom m3 is an excellent wireless VoIP phone with excellent battery life, very good range, and very good features. The multiple simultaneous SIP registrations is a huge plus. I wished the base station supported PoE, but it's not a big deal for home users since most home users don't have Power over Ethernet switches. I'll be interested to compare the snom m3 with the new line of Polycom KIRK wireless DECT SIP phones, but for now the snom m3 is my favorite cordless SIP-based VoIP phone!

Price:
You can buy the snom complete set (with base + handset) on Amazon for $172 , and an additional handset on Amazon for $142.

fring for iPhone has arrived! I'm a huge fan of fring, which I like to call the Swiss Army knife of VoIP/IM communications since fring works with AOL/AIM, MSN Messenger, Google Talk, Skype, Twitter, Yahoo! Messenger, and SIP registrars/IP-PBXs. I've used fring on my Windows Mobile 6.1 phone to connect to an Asterisk-based IP-PBX using SIP which enabled me to remotely make and receive calls. fring is a currently a free pre-release app free on iTunes.

VoIP using fring is of course restricted to WiFi connections - it won't work over 3G, but still cool nonetheless. Further, according to the apps description in iTunes you can IM over 3G, GPRS, EDGE, or WiFi, so you can use fring as your centralized IM application on your iPhone.

Features:
• VoIP (Voice) Calls over WiFi
• Instant Messaging
• Integrated dynamic contact list 
with real-time contact availability
• SIP integration
• Multiple Connection types

Download fring for iPhone here.

Ok, now my head is getting dizzy from the number of times Microsoft Windows Live Messenger/MSN Messenger has had outbound VoIP-to-PSTN calling (2006), then pulling outbound VoIP calling (early 2008), and then putting it back in. Also, I believe it was 2004 when the Messenger client used Net2Phone before they pulled the plug. Well, apparently outbound PSTN dialing using VoIP is back in!

Windows Live Messenger has now teamed up with Telefónica to offer VoIP services. Previously Net2Phone and Verizon have had exclusive deals with Microsoft's Messenger client.

When you click on Make a Phone Call you see the dialpad window and it explains you can sign up with Telefonica's Voype service to call directly from within Windows Live Messenger.

Telefónica's rates seem decent as compared to SkypeOut. For instance,Telefónica charges $0.014 per minute for the U.S. comparaed to $0.021 SkypeOut calls. Unfortunately, there is no dial-in (DID) capability equivalent to SkypeIn with Telefónica's service.

The service uses prepaid amount in dollars. Increments of $5, $10, and $20 are available and you can set it up to automatically recharge the account when it reaches a certain threshold. To use it you just need Windows Live Messenger 8.0 and above.

If Microsoft really wants to compete with Skype what they should do is partner with all the major SIP trunking service providers (Bandwidth.com, DIDX, Junction Networks, Packet8, etc.) and offer them all as a drop-down list within Windows Live Messenger for quick and easy configuration. After all, unlike Skype which is proprietary, Windows Live Messenger is based on the SIP protocol. Further, Microsoft could allow Windows Live Messenger users to manually enter their existing SIP trunking service provider account info, essentially making Windows Live Messenger a SIP softphone client able to make and receive calls. Microsoft could even do revenue sharing with the SIP trunking service providers.

Even better, Microsoft could offer the ability for users to enter in custom SIP credentials to use with the user's SIP-based IP-PBX! Since in this scenario the connection is direct to the IP-PBX no revenue sharing is required. Of course, since SIP is SIP, a user could simply go into manual mode, and enter in, for example, their Bandwidth.com SIP trunking info thus bypassing the drop-down list, connecting directly to Bandwidth.com and eliminating any revenue share Microsoft might receive.

However, Microsoft could restrict the manual SIP credentials entered simply by having a database of their SIP trunking providers' URLs or Microsoft could simply stick something into the SIP header which the SIP trunking service providers can parse and detect and then give credit/revenue to Microsoft for sending the call from Live Messenger onto their network. So many ideas,  I should write a book.

Adtran recently launched their IP 700 series of IP phones in late April. Adtran sent TMC Labs the IP 706 model, which supports up to 6 lines, but the 700 series also includes the IP 712 which is identical feature-wise but supports up to 12 lines. Each line can be configured to register with unique SIP proxy/registrar servers. This allows a different line for every line key on the phone. A line is called a multiple call appearance (MCA) type if it will be assigned to one or more line keys on the same phone. It is called a shared call appearance (SCA) type if the line is shared across multiple phones. This is not to be confused with SLA (Shared Line Appearance) which maps PSTN lines to buttons on all the phones. Of course you need to assign two lines with the same SIP credentials to two different lines (MCA) for full call handling functionality.

Like most if not all IP phones these days, the IP 706 supports 802.3af Power over Ethernet (PoE) as well as TFTP booting of firmware and configuration from a TFTP Server. The Adtran phone will connect to your TFTP Server (option 66 on DHCP server) and look for a file called adtran_[MAC address of Adran phone].txt. So for instance, for the IP 706 phone I tested, it looked for adtran_00a0c831593c.txt on the TFTP Server when the phone was booted.

The configuration files are pretty easy to figure out and sample files are available. For instance, one of the first things you'll want to do to configure any IP phone is to setup the dialplan. I was able to easily figure out how to setup the syntax for the Adtran dialplan, as seen here:

# DialPlanExternal is for realm GE line types and DialPlanPBX is for realm GP line types
DialPlanExternal |911|2-9]xxxxxx+T3|2-9]xx[2-9]xxxxxx|[0-1][2-9]xx[2-9]xxxxxx|011xxx+T3|xx+#
DialPlanPBX |911|9911|1-8]xxx|9[2-9]xxxxxx+T3|9[2-9]xx[2-9]xxxxxx|9[0-1][2-9]xx[2-9]xxxxxx|9011xxx+T3|*2-9]0123456789*]+T3|*1xx|#xx+#|xx+#|**xxxx

The web admin was pretty intuitive and can be used instead of a config file on a TFTP server. Here's a screenshot of a the web interface:


Want to specify a corporate directory? No problem. Just export a comma separated file containing your corporate directory, upload it to the TFTP server and then add this line to the Adtran config file:

SystemPhonebook adtran_phonebook.csv

I exported my Outlook Contacts to a CSV file, including first name, last name, company name, title, email, street, street2, street3, city, state, ZIP, country, mobile phone, home phone, FAX, with column/field headings in the first row. The IP 706 will read the first row to automatically map the contact data into the system phonebook. Once imported, you can scroll through the system directory using the 4-way navigation button. Holding the up/down arrow doesn't cause it to auto-repeat. Fortunately, you can press the left or right arrow to page up/down through the contacts. The 4-way navigation button also acts a shortcut buttons. When on the home screen you can press one of the four directions to access Incoming calls, Missed Calls, Placed calls, and the Personal address book. The detailed contact details is pretty cool, especially since most phones only store name and/or company and the phone number.

Defining buttons is pretty easy. Here's some examples from my config file:

Button.1.Label Line 1
Button.1.Type line
Button.1.Line 0

Button.2.Label Line 2
Button.2.Type line
Button.2.Line 0

Button.4.Label x149 Tom
Button.4.Type speed
Button.4.Number 149

Button.5.Label DND
Button.5.Type DND

Button.6.Label vm
Button.6.Type speed
Button.6.Number 8555

Although the Adtran IP700 series was probably designed initially to work with the Adtran NetVanta 7060 and 7100, the Adtran IP700 series are SIP-based so the phones work with any SIP-based IP-PBX. I was able to register the phones on the Asterisk-based trixbox platform very easily. Once registered, I was able to make calls to Aastra and Polycom IP phones. The voice quality on both ends seemed very good. Usually the sound quality when using a handset is not an issue for any IP phone - it's when you try and use the speakerphone that sound quality issues arise. You need good echo cancellation to make sure the remote speaker's audio isn't fed back into the speakerphone. Polycom is renowned for their superior sounding voice quality in speakerphones, however, I was pretty impressed with the sound quality on the Adtran IP 706 when in speakerphone mode. The speakerphone volume when set to maximum is extremely loud and without any distortion. I doubt even in the largest of conference rooms that the loudest volume setting be required, but it's good to know it has the capability.

Overall, I like the button feel. not too hard, not too soft. Navigating the menus and options was very intuitive, though there is no key auto-repeat, which would be handy when scrolling quickly through the built-in directory book. Though, as I previously stated, you can use the left or right arrow to page up/down. The LCD was excellent - it's very bright and uses icons to indicate various features. For instance, a bell indicates your phone will ring, while an 'X' through the bell indicates DND mode. Similarly, a phone icon displays next to each line with or without an 'X' depending on if the line was registered with the SIP registrar or not. A U-turn arrow indicates a line is being forwarded. An envelope displays at the top of the phone if you have voicemail, along with the number of new messages. The phone has a slightly slow boot-up time taking 83s to fully boot. Comparatively, an Aastra 57iCT took 53s and a Polycom IP650 took 65s. Not a big deal, since you don't typically reboot your IP phone.

The Adtran IP phone supports busy lamp fields (BLF) using the Broadsoft method not the Sylantro method. This may be important if you are deploying Asterisk, since Asterisk only supports the Sylantro method. Personally, I have no need for BLF on our Asterisk-based IP-PBX, and no one in our office uses BLF, but certainly receptionists might find BLF useful. Other than the BLF feature, all other features worked on the trixbox system I was testing it with.

I was able to make outbound hands-free auto-answer intercom calls from the IP 706 to an Aastra phone. First I had to define the star code (*74) for initiating hands-free intercom calls. From the IP 706 I simply pressed the HFAAI (hands-free auto answer intercom) button on the LCD display under the More menu and dialed an extension which will immediately cause the remote phone to ring off-hook into hands-free speakerphone mode. You can also setup a speed dial for HFAAI so you don't have to go into the More submenu - a two step process.

Although outbound HFAAI calls from the IP 706 work, I wasn't able to get the Adtran phone to receive hands-free intercom calls from an Aastra phone. For instance, I made a from x149 Aastra phone to the IP 706, and although the IP 706 LCD displayed "Intercom - 149" it rang normally and did not go off-hook into speakerphone. I have to lift the receiver or press the speakerphone button to answer the call. I contacted Adtran technical support and they were quickly able to determine the issue. The phone responds to "alert-autoanswer" or "autoanswer" in the SIP header, so it's possible to tweak Asterisk to get it to work.

For speed dials, the Adtran IP phone supports 100 Personal and 300 System entries, no matter how many fields are in each record. You can even enter in pauses for speed dials with a "P" for a 2 second pause, useful for dialing through auto-attendants to an extension (i.e. 98005551234PP100).

In addition, you can export Outlook Contacts into a CSV file and put the CSV file on the TFTP server, which will be the global (not personal) system phonebook. You can also import a .CSV file directly to the phone via the phone's Web interface for your own personal phonebook and speed dials. The personal contact directory can be imported from the personal web GUI. You log into http://x.x.x.x/admin for the admin GUI, but just log into http://x.x.x.x for the user GUI.  It allows for the upload (append or replace), and backup of the personal directory.  The format is the same as the System Directory csv file.


Users can even enable call forwarding from the phone's web configuration. This is useful for when the IP-PBX doesn't support call forwarding. It even supports forwarding to an outside number.

From the phone itself you can test the audio of the handset speaker and the phone speakerphone. You can set the input to the handset microphone and have the output directed to the handset speaker or the speakerphone. Further you can test the button LEDs by turning them all on and you can test the LCD on the phone. Adtran claims that the IP700 series draws less than 6.49 watts of power under normal operating conditions. I was going to test it with my Kill a Watt electric meter, but I seemed to have misplaced it.

One nicety is you can modify the splash screen simply by downloading a 216x336 pixel 16-bit bitmap file to the parameter IconPixmap. This might be useful for OEMs or even IP-PBX vendors that want to do branding.

On inbound calls, the blue Messages light flashes, which is the button used to check your voicemail. You can't press the flashing Messages button to answer the call on speakerphone mode. I would prefer that it flash the speakerphone button instead. The reason is that when I first hooked it up and called it for the first time, I instinctively pressed the Messages button since it was flashing and I wanted to answer it via speakerphone mode. A minor complaint for sure.

Another test I performed was redirecting an inbound call to voicemail. You have a couple options. First, you can simply click 'Ignore' on the LCD and that will simply mute the ringing, but the caller has to wait until the ring duration setting has been met before going to voicemail. The proper way is to press the 'Vmail' icon on the LCD which will redirect the caller to the voicemail system. When I first attempted this, it sent the caller into the voicemail logon asking the caller for their extension and password. After perusing through the Admin Guide, it seemed like I had the voicemail settings correct. But then I realized I needed to do a call transfer direct to voicemail (*86 code) to the phone's extension (135). So I needed the *86 code. I simply needed these two lines in the Adtran config file:

MessagesCallback 8555   # For 1-button access to check voicemail
Reg.0.Voicemail  *86135 # For redirecting callers to voicemail.

The phones include an adjustable desk stand or can be wall mounted. An integrated headset jack with electronic hook-switch eliminates the need for a mechanical handset lifter. The electronic hook switch is compatible with GN Netcom and Plantronics headsets.

Features:

• Adaptive jitter buffers and packet loss concealment algorithms
• Six programmable buttons
• Large backlit display, with 6 rows by 35 characters (IP 706), 9 rows by 35 characters (IP 712)
• Message waiting indicator
• Four-way navigation
• 802.3af Power over Ethernet (PoE)
• Integrated headset jack
• Distinctive ring tones by number
• Multiple call appearances
• Three-way conferencing
• Busy Lamp Field (BLF)
• Shared Line Appearance (SLA)
• Hands-free auto-answer intercom
• Distinctive incoming call treatment/call waiting
• Visual ringing alert/message waiting indicator
• Voice activity detection and comfort noise fill
• Full-duplex speaker phone
• Three-way conferencing
• G.711u, G.711a, G.729A (Annex B)

Ratings Score Installation Documentation Features Usability Performance Overall

Pricing: The Adtran IP 706 is $249 and the Adtran IP 712 is $299.

Conclusion
I like the aesthetics of the IP 706. It's a nice clean design with a bright LCD and it has a very intuitive navigation menu on the phone. Similarly, the web interface was easy enough to navigate and figure out. The adaptive jitter buffers and packet loss concealment algorithms are a nice addition to ensure voice quality. A way of importing personal contacts into the phone itself via the web interface would be nice, but I do like that the Adtran speed dials support pauses - not all IP phones do, which makes them less useful when dialing auto-attendants with extensions. Overall, I was pretty pleased with the Adtran IP 706's style, performance, and features. Customers have yet another choice when choosing a SIP-based IP phone. Watch out Aastra, Grandstream, Linksys, Polycom, and Snom - there's a new IP phone in town!

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.

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:

I discovered two Microsoft job postings several weeks ago and have been meaning to blog the discovery that Microsoft Office Communication Server R2 will ship in December of this year. One is for 'Software Development Engineer in Test - MBD - RTC' and the other for 'Lead Software Develpment Engineer in Test - Office Communications Server'. One of the job posting says:

"The Office Communications Server team is starting a new deployment/management/administration team at Beijing. The Office Communications Server product team must deliver both an on-premise version of the Server as well as offering a hosted Service for Small and Medium Businesses. The test lead needs to understand current infrastructure (MMC, Setup.Exe, MOM, etc.) to help out in next release called wave13 due to ship in December 2008.

As the team ramps up in MBD China, the team gain experience in these areas and can form opinions/ideas as to how best to architect a new solution for the future. The test lead will work with PMs and Devs to identify scenarios, architect and build a new solution for both on premise and hosted offerings due to release in Q2 2010 for deployment/administration/monitoring. The new team would to build a new infrastructure that is conducive to hosted environments.

Examples include: web based flavor of management tools, a topology builder that can visually display a topology and once the solution/topology 'compiles', settings can be rolled out across the server farm (as opposed to admin's having to run setup.exe on every machine out there). Monitoring also needs to be addressed in such a way that the server or service can be somewhat self correcting when alerts fire notifying administrators of a problem.

Key Areas of Responsibility:
Attract, Lead, Train, mentor, grow, and retain SDET talent at all levels
Work closely with the program management and development teams to drive quality through design and implementation
Participate in product spec reviews, design, triage, scheduling, and other product development process
Develop comprehensive test plans assuring the overall quality of the project, including functionality, security, performance and scalability
Hands-on writing test cases and test code
Lead the SDET team in implementation of a scalable, efficient test automation strategy
Work closely with program management and development teams to ensure appropriate quality metrics and goals are defined and tracked throughout the project lifecycle
Drive quality criteria for release and signoff
Work with other SDET leads and managers to ensure engineering excellence initiatives are driven effectively across the Commerce Platform Group."

Apparently, Beijing, China will host some core Microsoft OCS 2007 R2 folks instead of Redmond, Washington in the U.S. I'm not sure how I feel about that. Seems like American IT jobs are constantly being outsourced. But I guess it's a global economy and I'm certainly not for protectionism either. Just very sad that IT jobs are outsourced because labor costs are less expensive than in the U.S.

One very fascinating part of this job posting is where it says, "architect and build a new solution for both on premise and hosted offerings due to release in Q2 2010 for deployment/administration/monitoring. The new team would to build a new infrastructure that is conducive to hosted environments."

It would appear Microsoft wants to take OCS 2007 into hosted environments by 2010. Many service providers offer very successful hosted Exchange 2007 services, so offering hosted unified communications (IM, VoIP, video, collaboration, etc.) via a hosted OCS 2007 offering is a natural progression. Hosted OCS 2007 is a much higher value proposition than hosted Exchange email services and could be a boon to service providers. One advantage of a hosted OCS 2007 offering is that it removes the complexities of deploying OCS 2007 in the enterprise.

So look for a hosted OCS offering in 2010. You heard it here first!

I'm not going to go on another rant blasting blog aggregators or websites that steal other people's content (aka sploggers). However, I came across one website that took my content, re-worded one sentence and tried to claim it as its own. The change is so laughable I busted out laughing!

First, my blog entry titled Court Bans VoIP App on iPhone was one of the first if not the first U.S.-based news outlet to talk about how a German court banned the sipgate VoIP application on the iPhone. In the article, I wrote:

Apparently, the court felt that sipgate would "lure" iPhone users into "jailbreaking" their iPhones. WTF? Banning software because it might entice customers to do something bad? Glad I live in the good ole' freedom-loving USA where we punish people for committing crimes not for "intentions" of committing crimes, jailbreaking iPhones, violating terms of service contracts, etc.

Now compare that with the scammer over at Unwiredview.com which posted the story as though it were their own:

Apparently, the court felt that Sipgate would lure iPhone users into jailbreaking their iPhones. WTF? Banning software because it might entice customers to do something bad? Glad I live in the good ole' freedom-loving India where we punish people for committing crimes not for "intentions" of committing crimes.

It's virtually identical -- even copying my WTF outrage. However, simply change good ole' freedom-loving 'USA' to 'India' and presto-bango, you've got yourself a brand-spanking 'new' news article! I couldn't help but laugh at this.

Obviously, there are tons of automated blog and news site aggregators out there and I've learned to just ignore them. But in this case, someone actually took the time to HAND EDIT it and take out 'USA' and change it to 'India'. Either they're anti-USA or just very pro-India. Regardless, a pretty stupid edit job, but definitely made my day seeing such a half-ass edit job.

On a related note, a major news outlet, CBC News (Canadian Broadcasting Channel News) mentioned my story writing, "Sipgate's Thilo Salmon told the VoIP and Gadgets blog the company was not allowed to argue its case before the Hamburg court prior to the injunction being issued" but alas they didn't actually link to my story.

Very annoying. It's not that I need the Page Rank from the inbound link, it's just common courtesy to cite & link the source of the article.

This news about American troops using VoIP & video is a few weeks old, but I wanted to share it, since it's pretty interesting. SightSpeed VP of Marketing Eric Quanstrom mentioned the news to me awhile ago when we were on a SightSpeed video call. Using SightSpeed, U.S. troops are able to play PlayStation, XBox and the WII against professional athletes (Philadelphia Eagles) while seeing them (video) and "trash talking" them over the Internet (VoIP). Pro vs. G.I. Joe online gaming using SightSpeed for the communications - pretty sweet! SightSpeed has an advantage over Skype in that it supports up to 9-way video, while Skype is limited to 2-way video.

Check out the news...

'PRO VS. GI JOE' COMES TO PHILLY

SightSpeed Video Conferencing Connects Pro Athletes with Troops Stationed Abroad

SightSpeed Inc., the leading provider of Internet video chat and conferencing, will connect sites in Philadelphia and Kuwait, Tuesday, Sept. 16, when three Philadelphia Eagles players square off eye-to-eye against U.S. Army troops in a friendly Pro vs. GI Joe interactive online Madden NFL '09 battle.

Pro vs. GI Joe is a nonprofit organization that manages and sponsors real-time video game competitions between professional athletes in the United States and troops stationed abroad over the Internet using PlayStation, XBox and the WII. The organization - whose motto is "Doin' A Little for Those Who Do A Lot" -- also arranges for the troops' family members and loved ones to travel to the U.S. sites to be a part of the event.

SightSpeed will provide the live, real-time video link between a USO center in Kuwait and the Eagles' NovaCare practice facility in South Philadelphia at 5 p.m. EDT and midnight in Kuwait.

The NFL pros - LB Stewart Bradley #55, TE Brent Celek #87 and DE Chris Clemons #91 - will be able to see and "trash talk" with their Army competitors in the Middle East through SightSpeed monitors, which will be positioned side-by-side with TV monitors displaying the video-game action.

The GI Joes -- PFC Justin Gindhart, 20, of Philadelphia; SPC Joel Dolliver, 22, of Londonderry, N.H.; SPC Steven Moore, 22, Mahaffey, Pa.; PFC Jeff Holt, 29, of Coleman, Mich.; and PFC Marcus Peden, 20, of St. Louis.-- are from the Army's 10th Mountain Division currently stationed in Baghdad, Iraq. They will be traveling to the USO Center in Camp Virginia, Kuwait.

Following the Madden NFL '09 matchup, families of the troops will be able to reconnect with their loved ones stationed far away via SightSpeed.

"This is our way of giving back to the people who are risking their lives for us every day," said Greg Zinone, founder and president of Pro vs. GI Joe. "We've found a way through technology to give our troops some temporary relief from their duties and briefly reunite them with their families while abroad."

SightSpeed is portable and connects people anywhere in the world. It's an easy-to-download software application - free to consumers -- that provides full-color, full-motion Internet video connectivity to laptop or desktop computers, regardless whether users are running Windows, Mac or Linux platforms. And thanks to SightSpeed's ability to record video chats, the players, their families and their teammates will have digital video records of their historic matchup.

"SightSpeed is the perfect connection for Pro vs. GI Joe," said SightSpeed CEO Peter Csathy. "Our quality is outstanding and we take distance out of the equation. The interactions we've seen so far between the pro athletes and the Army troops have been outstanding. The players on both sides can see themselves clearly and they have a lot of fun. And for us, it's tremendously uplifting to give the families of our service members the opportunity to see and chat with their sons or daughters on duty so far from home."

About Pro vs. GI Joe
Pro vs. GI Joe is a nonprofit organization that boosts the morale of our brave military men and women by setting up real-time video game competitions between professional athletes and troops stationed overseas via the internet. Pro vs. GI Joe is an official partner of the USO and is the first organization to bring live gaming to troops in the Middle East. The organization was founded and run by Greg and Addie Zinone, based in Fairfax, Virginia. Addie is a Staff Sergeant in the US Army Reserve and has served two tours in support of Operation Iraqi Freedom. Her husband Greg came up with Pro vs. GI Joe because he believes if you don't serve, you have an obligation to support those who do. To find out more about Pro vs. GI Joe, please visit www.provsgijoe.org.

Polycom today announced the launch of its latest KIRK Digital Enhanced Cordless Telecommunications (DECT) wireless products. Polycom has introduced three new products: the KIRK Wireless Server (KWS) 300, the KWS 6000, and the KIRK 5040 handset. which are all SIP-based.

The name KIRK certainly evokes Captain Kirk from Star Trek and most likely intentionally, since Captain Kirk and his crew made the wireless communicator famous 40 years ago. No doubt Polycom has some Trekkies in their engineering or marketing teams.

Of course, I should mention that Polycom acquired Spectralink, a well known wireless phone manufacturer and Spectralink previously acquired Kirk Telecom, the makers of these DECT wireless phones. So the product name is simply a reflection of their corporate name. So perhaps all this Trekkie analogy business is moot. Or maybe they named their company Kirk Telecom to honor Captain Kirk. Stranger things have happened.

All KIRK solutions are scalable, both in terms of the number of users as well as the coverage areas supported. The latest additions to the KIRK Wireless Server portfolio include:

• The KIRK Wireless Server 300, a SIP-based wireless telephony system, is ideal for smaller sized businesses, by scaling support from one to 12 handsets The KIRK Wireless Server 300 is a single-cell solution that can support up to four simultaneously calls and up to six KIRK repeaters in order to extend the coverage area. Each KIRK repeater increases the coverage area by approximately 50 percent.
   
• The KWS 6000 is a SIP-based enterprise wireless telephony solution that scales from just a handful up to more than 4,000 users. Up to 256 radio units are supported, which when combined with the KIRK Media Resource, can support more than 1,000 simultaneous calls. Each KIRK base station handles 12 simultaneous calls, and customers can scale up based on their individual needs. Additionally, KIRK repeaters can be added to increase the coverage area by approximately 50 percent.

The KIRK 5040 handset, the newest addition to the KIRK product line, is a lightweight DECT phone that combines an intuitive user interface and wireless headset that can be operated hands-free and wirelessly with a Bluetooth headset. Like the KIRK 5020, the 5040 can quickly be switched to silent mode and will distinguish between external and internal calls by ring tone. The KIRK 5040 handset also features an intuitive user interface and a large color-display offering an experience similar to a mobile phone and with the added benefit of hands-free operation.

Features of the 5040:

• TFT colour display (65.000 colours, 8 lines of text/icons)
• Li-ion battery
• 4 Way navigation key
• 2 Softkeys
• CLIP (40 caller-ID presentations)
• Date and time in display when supported by system
• Internal/external ring pattern
• Volume control
• Telephone book with 250 name entries (4 numbers per name)
• Auto login - roaming between 10 different installations
• Silent mode (mutes all alerts/calls)
• Alerting on silent mode (choice from display flash, vibrator or short ring)
• Call list of incoming/missed/received (last 40 entries)
• Redial function from call list
• Speed dial
• Auto answer with different settings (after 1st ring/when lifted from charger/on headset/loud speaker on)
• 10 different ring signals and adjustable ring volume
• Key lock
• Auto key lock
• Vibrating alert
• Any key answer
• 11 menu languages (UK, FR, DE, ES, IT, NL, CZ, PL, DK, NO, SE)
• Headset connection
• Ring signal in headset
• Adjustable volume in headset
• Answer/end calls via headset button
• Microphone mute
• Speaker on auto-answer
• R-key for transfer and special services
• Adjustable alerting volume (low battery/low coverage/incoming message)
• Adjustable backlight delay (for max. battery conservation)
• Text messaging - max. 72 characters per message (system dependant) 10 user defined messaging templates
• Stores 20 messages
• Speech/stand by time: Up to 15/100 hours
• Weight incl. battery: 110g
• Size (LxWxH): 146x48x19mm
• 2 types of chargers (w/wo USB 2.0 connection)
• Suitable for Bluetooth headsets

Pricing & Availability
The new KIRK solutions are available worldwide through Polycom's certified reseller partners. The list price for the KWS 300 is U.S. $360. The KWS 6000 list price is U.S. $1,200 and includes a server and one base station, which supports up to 30 users. With the scalable nature of the KWS6000 it can also be set up for more users. The KIRK 5040 handset sells at a list price of U.S. $310. To learn more about Polycom's KIRK phone solution, head here.