Recently in VoIP Category

verizonlogo.gifRich Tehrani writes a great story with some historical context on how Verizon lost a VoIP patent case (on all 6 patent counts) to Cox Communications. Unfortunately, a bit too late for Vonage which paid millions to settle their patent case with Verizon.
Really though the large telcos are the winners here and consumers are the losers. The US patent system continues to be a barrier to true innovation and consumers are being hurt -- severely so in some cases -- by large companies who use the patent system to prevent other companies from succeeding. Instead of competing with better technology alone, these large companies use large amounts of patents at once to scare new entrants into submission. Hopefully, in the IP communications space, Cox Communications will mark a point in time when large companies slow down their IP communications patent infringement onslaught.
Check out the full story.

snom m3 review

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snom m3 with base station
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
             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.

snom m3 advanced settings
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.
snom m3 telephony settings.jpg

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.

snom m3 configure identity

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.
digium-asterisk-world.jpgI've had some big news I had to keep under wraps until today. My company, Technology Marketing Corporation (TMC) and Digium, the Asterisk Company, today announced that they have partnered to host Digium|Asterisk World™ during INTERNET TELEPHONY Conference & EXPO East 2009 in Miami, Florida. Having ITEXPO host Digium|Asterisk World is a huge win for TMC and further cements TMC's ITEXPO as the most preeminent VoIP and IP communications show. In fact, I recently came across this quote from Phonevite, "With the demise of the VON Show, the Internet Telephony Conference & Expo (aka ITEXPO) is now regarded as the biggest and most eminent show in the VoIP industry."

Digium|Asterisk World at ITEXPO will be the conference that addresses "Everything Asterisk" for business users, resellers and executive decision-makers. The event will be held February 2-4, 2009 at the Miami Beach Convention Center.

All can come to Digium|Asterisk World to discover how Asterisk, the world's most widely used open source telephony software, can save them money and allow them to create more flexible telephony solutions.

"We are honored to host Digium|Asterisk World, as open source technology is such a critical driver the growth of the communications market," said Rich Tehrani, TMC president and ITEXPO East 2009 conference chairman. "Mark Spencer and the Digium team are without question true pioneers and innovators. What is important to note is that by joining forces with Digium|Asterisk World, ITEXPO reinforces its position as the industry's most valuable, major communications event worldwide."

The conference will include both booth exhibition space and a Presentation Theatre on the EXPO floor. In addition, TMC and Digium will collaborate to create the conference track agenda, which will be announced in the coming months.

"Digium has led the Asterisk revolution in telecommunications," said Mark Spencer, founder and CTO of Digium and original creator of Asterisk. "Hosting Digium|Asterisk World at IT EXPO East 2009 will allow us to share the vision and power of Asterisk with a broad set of customers who might not be familiar with open source. By giving them their first taste of Asterisk, we empower them to not only save money but to use and create new technologies that never existed before."

ITEXPO East 2009 is the world's largest and most significant communications technology event, featuring more than 200 companies exhibiting on the EXPO floor and hundreds of sessions led by the industry's most prominent thought leaders. The show helps attendees identify the issues and challenges affecting the deployment of communications technologies. It provides a comprehensive forum for evaluating the latest products and services and delivers a face-to-face networking opportunity that service providers, carriers, resellers, distributors, equipment manufacturers and IT executives from enterprise and SMB companies need to cultivate new business relationships.

For additional information on ITEXPO East 2009 or Digium|Asterisk World at ITEXPO, please contact TMC's Dave Rodriguez at 1-203-852-6800, ext. 146 or at drodriguez@tmcnet.com.

fring Adds VoIP to iPhone

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fring-itunes.jpg
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 fring-iphone.jpg 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.
windows-live-messenger-make-phone-call.jpg

windows-live-call.jpg 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 IP 706 Review

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adtran-ip-706.jpg
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:
ip-706-web-admin.jpg

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.
ip-706-import.jpg

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.
motorola-femtocell-voip-prototype.jpg
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.
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:


More on Skype for Asterisk

| 5 Comments
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-for-asterisk.jpg
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