Kerry Garrison and the team at VoipStore.com have posted another great product overview, this one is on the improved analog cards from Digium.

The Digium® TDM410 is a half-length PCI 2.2-compliant modular gateway card for connecting analog telephone stations and analog POTS lines through a PC. It supports a combination of up to four station or trunk modules for a total of 4 lines.

Using Digium’s Asterisk® software and standard PC hardware, one can create a telephony environment that includes all of the sophisticated features of a high-end business telephone system.

Using an industry-standard bursting, bus-mastering interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus™ technology, the TDM410 eliminates the requirement for external gateways, with industry-leading performance and price. The trunk and station modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces. Scaling of this solution is accomplished by adding additional TDM410 or other Digium analog interface cards.

Digium Analog cards, with there interchangeable single and quad FXS and FXO modules can eliminate the requirement for seperate channel banks or access gateways. Digiums commercial, toll-free quality High Performance Echo Cancellation (HPEC) software is available to our analog customers at no additional cost. The optional VPMADT032 hardware echo cancellation module provides the same toll-quality HPEC but without the perfoamce inpact of a software based solution. Scaling of an analog card solution is accomplished by adding additional cards and FXO / FXS modules.

Core Features

• 4 Ports for connecting analog telephones or POTS lines
• Half-length Analog Card
• Up to 4 FXS or FXO Modules
• High Performance Echo Cancellation (HPEC) Software (Optional)
• TDM410 for use with a PCI 2.2 compliant slot

888VoipStore.com: Digium Products Page

Kerry and the team at VoipStore.com have put together this great overview on the Digium TDM800P Analog cards.

The Wildcard TDM800P is a half-length PCI 2.2-compliant, 8 port card for connecting analog telephones and analog POTS lines through a PC. It supports combinations of FXS and/or FXO modules for a total of 8 lines.

The TDM800P provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single PCI bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the TDM800P reduces part complexity, cable clutter, and points of failure.

The TDM800P contains two module banks. Each bank supports up to four analog interfaces. The module banks may be filled either with one standard Digium® quad analog module (FXS – S400M, FXO – X400M), or up to two standard Digium® single analog modules (FXS- S110M, FXO – X100M) enabling the creation of any combination of ports

The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both FXO and FXS interfaces.

If the hardware echo cancellation module is not installed, the TDM800P may be used in conjunction with Digium’s High Performance Echo Cancellation (HPEC) software, a commercial and toll quality hybrid echo cancellation solution. Digium’s HPEC provides 16ms to 128ms of selectable near-end ITU G.168 compliant echo cancellation in software.

Using this card in concert with Digium’s Asterisk® software, standard PC hardware, and the Linux® OS, you can create SME or SOHO telephony environments capable of satisfying the needs of small or medium business applications at an industry-leading price.

Digium Analog cards, with their interchangeable single and quad FXS and FXO modules can eliminate the requirement for separate channel banks or access gateways. Digium’s commercial, toll-free quality High Performance Echo Cancellation (HPEC) software is available to our analog customers at no additional cost. The optional VPMADT032 hardware echo cancellation module provides the same toll-quality HPEC but without the performance impact of a software based solution. Scaling of an analog card solution is accomplished by adding additional cards and FXO / FXS modules.

• 8 RJ-11 interfaces on a single PCI bracket
• 8 Ports for connecting analog telephones or POTS lines
• Half-length Analog Card
• Up to 4 Single FXS or FXO Modules, or 2 Quad FXS or FXO Modules
• High Performance Echo Cancellation (HPEC) Software (Optional)
• VoiceBus™ technology

888VoipStore.com: Digium Products Page

What this phone is
The SNOM M3 is a wireless phone that uses the DECT protocol versus being a WiFi phone. There are tons of advantages of DECT technology including call quality, battery life, and range. DECT also allows for multiple handsets to register to the same base station and the M3 takes full advantage of this by allowing each handset to either share an extension or act as independent extensions, or any combination. In some cases you may want a single extension to ring on multiple devices or multiple users can all have their own extension with up to eight handsets being able to register to each base station.

Features
The M3 has a nice LCD display that isn’t too difficult to read and has volume controls, speakerphone toggle, and a headset jack on the side. Each handset comes with it’s own charger so you can conveniently place the chargers near where you are going to use the phones.

• 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

Setup
The setup of the M3 is fairly simple as there isn’t anything to connect besides ethernet and power. Once connected, press the volume up control on the handset and the unit will give you the IP address of the base station. Browse to the IP of the base station and setting up the base to connect to an IP PBX is the same as any other SIP device. I tested the unit with both trixbox CE with the updated Endpoint Manager that is designed to work with the M3 and I manually configured it to work on trixbox Pro where I have been using it as a daily phone for some time now. The only thing that is really different about the M3 is that it can support up to eight SIP registrations (extensions), then for each handset so can specifiy which extension rings which handset and what extension information to use for outbound calls on each handset.

Usage
There is nothing really different about the M3 than any other wireless phone on the market in terms of how it works, its a phone after all, its makes and receives phone calls and does it well but is it any better than other wireless phones? The two most common types of cordless phones in use are analog cordless phones attached to an ATA or a WiFi phone connecting to your wireless router. Since that is the market segment the M3 is up against, let’s use that as the comparision.

Range
I have a decent 2.4ghz cordless phone, with this phone I can walk out the front door and get right to the sidewalk in front before starting to lose signal. A Wifi phone will get me about another 30 feet or so before it starts dropping. The WDECT handset that comes with the Aastra 480i CT will let me walk down the street about one house in each direction. The M3 with the full DECT implementation will get me about 3 houses in each direction. As far as range goes, the M3 is a clear winner.

Battery Life
WiFi is absolutely horrid for cordless phones as there is no real standby mode so the radio is at full power 100% of the time. The different WiFi phones I have tried basically have to sit in a charger so that they are always ready to use. The M3 handset goes without a charge for 4 – 7 days depending on usage. Again, the M3 is a clear winner.

Where does it fail?
The M3 is not perfect and could certainly be improved and some improvements have already occured since I got my original unit. While I am happy with the call quality and the speakerphone, the overall build quality feels a bit cheap. The original units had a fake antenna which I found to be a bit cheesy, this has since been removed. If there is one feature I think is really a shame that the M3 doesn’t have, its power over ethernet. With PoE support it would be easier to throw the base station up into the hanging ceiling in an office to provide optimum coverage without having to run power. A nice, but not necessary feature would be an ethernet passthrough port. The second port I can live without but the M3 really should support PoE.

Results
For around $240 you get the base station and one handset and additional handsets run about $110 each so after you put two or more handsets onto the system the price is less than most other SIP phones. The call quality is good, the speakerphone works well, the range is terrific and the battery life is awesome. Sure it could use some refinements but overall its a really solid device with a good set of features. The final testement to what I think of a phone is where it ends up after the review period is done. There is a shelf in the garage loaded with every phone on the market that I could choose to use in my office or home. While the M3 is not my primary desk phone it is on the network as a permenant extension so that I have wireless ability and it is part of my FindMe so that I can work anywhere in the house. The M3 may have some faults but the overall result is that the pros outweigh the cons and are enough that I would highly recommend this phone to anyone looking for a wireless sip phone solution.

SNOM
http://www.snom.com

Overview
Like the SPA-941, the newer SPA-942 is a SIP compliant VoIP phone aimed squarely at the business market. Both phones combine the technology originally developed by Sipura, combined with the look and feel of the Cisco phones to create a mid-range device that is affordable enough for small businesses and yet good enough for the most demanding companies. The phones support the SIP standard well enough to be able to be used on any of the popular IP PBX systems including Asterisk, Call Manager Express, SipX, PBXNSIP, and others as well as being extremely easy to provision with Linksys’ own SPA-9000 IP Key System.

Virtually every thing that people disliked about the old Sipura SPA-841 has been addressed and improved upon from the handset weight, the keys, the display, the overall style, and the sound quality. With the SPA-942, the few complaints about SPA-941 such as lack of backlit display, single Ethernet port, and lack of power-over-Ethernet have all been addresses. Actually, finding something to complain about with the 942 is actually nothing more than a wish list for a future phone.

Features
The SPA-942 is a full function SIP phone with 2 line call appearance (4 line optional) with an easy-to-use menu system controlled by the 4 way directional pad. Four softkeys are used to control different functions based on what screen you are on.

SPA941 Hardware Features:

• (2) RJ-45 100BaseT Ethernet Ports
• High Quality, Hi-Resolution Pixel Based Display
• 128 x 48 line display
• Four Line Keys
• Four Soft Keys
• “Solid” Handset
• Headset Port 2.5mm
• Full Duplex Speakerphone
• Interoperability with Asterisk and other SIP based platforms
• Excellent Voice Quality
• Secure Calling via sRTP
• Network Based Ring Tone Support
• Call Transfer, DND, Conferencing, Call FWD, etc.
• SIP B and Bridged Line Appearance Support
• Remote Provisioning via HTTPS, HTTP, TFTP
• Support G177u, G711a, G726, G729a, and G723 codecs
• Message waiting indicator light
• Handset, Headset, and Speakerphone Gain controls
• VLAN ability
• Backlit display
• Power Over Ethernet Capability

When ordering the SPA-942, keep in mind that the phone does not come standard with an AC adapter since this version of the phone supports power-over-ethernet. The SPA-942 is easily configured from the built-in web interface or it can be provisioned through XML files from a tFTP site. When used with the SPA-9000 key system, the phone is automatically setup as well.

For those who read my review about the SPA-941, my only complaints where the lack of a backlight on the display, only a single network jack, and no PoE (power over Ethernet) support, the SPA-942 addresses all three of these issues. Linksys listened to us and only about 15 of the SPA-942s ever shipped without a backlit display (does this make those units collector items?). Adding the second network jack saves you from having two network jacks at each person’s desk, and PoE support is a welcome plus for enterprise installations.

Usage
Let’s cut to the chase here, if you are looking for a quality business phone at a reasonable price, this is it. My company has installed around 150 of the 941/942 phones for different clients and everyone just loves them. The ONLY issue is that some people want some programmable keys for seeing the status of another line (like Asterisk’s BLF function) like the Grandstream GXP-2000 or the SNOM 320/360 phones. To put everything into perspective, I could have any phone made on my desk and I have an SPA-941 at my office and an SPA-942 located remotely at my house (which happens to be using a Linksys Wireless G Bridge, more on this later)

Initially the SPA-942 will configure itself via DHCP and then you can do the setup via the web interface which works quite well. When trying to provision 50 at a time, it is much easier to create config files for each phone and then a DHCP server that can push out DHCP Option 66 (Boot Server Address).

In practice, the phone has superb sound quality and is quite easy to use with an intuitive context-sensitive menu and soft buttons for the different functions. When the handset or any key is pressed, or when the phone rings a very smooth and bright backlit lights up the display completely. In typical fashion, this is one of the nicest backlights on any monochrome LCD we have seen so far. It would be nice if the timeout could be adjusted, or an option to simply leave the display on all the time would be even better.

The SPA-942 is fully SIP-compliant so it is pretty simple to configure to use with most VoIP services that support the SIP protocol including Asterisk, Broadvoice, Teliax, Gizmo Project, etc.

Summary
When part of your job is to review, test, and certify phones you can have phone you want on your desk so it stands to reason that the phone that sits on my desk is as good as it gets. This is precisely why the SPA-942 is the phone that sits on my desk.

The 942 is testament to how Linksys listens to the clients (and especially reviewers) and added the features to the 942 that were lacking in the 941 (namely the backlit screen and PoE support).

Linksys
http://www.linksys.com

Features
The Polycom Communicator C100 is a USB powered speakerphone for use for softphones on Windows XP based systems. The front of the device features call pickup/hangup key, application launch key, volume adjustment keys and a mute button. When muted, the ring around the keys glows red.
The unit features dual high frequency response cardioid microphones, a 3.5mm headphone jack and built-in USB cable. The Communicator features automatic gain control and dynamic noise reduction.

• Size: 5.25” x 3.24” x 0.85”
• Weight: 5.4 ounces
• Interface: USB 1.1
• Call Pickup/Hang-up key
• Application Launch key
• Volume Up/Down keys
• Mute key
• LED light-ring for indication of call status
• Speaker Freq. Response: 300 Hz to 19 Hz
• 2 cardioid microphones 200 Hz to 10 kHz
• 3.5 mm headphone jack
• Automatic Gain Control
• Dynamic Noise Reduction
• Gated microphones with intelligent switching
• Full duplex operation
• Carry Case

Setup
There really isn’t much setup to do besides plugging the device into an available USB port on your computer. Under Windows XP, the Communicator will automatically be detected as a USB sound device and is then ready for use with any software that uses standard Windows sound calls. This means that the Polycom Communicator can also act as a sound card for PC games and is a great microphone for recording audio such as podcasts.

With the integrated USB cable, you will never be stuck looking for a lost cable again, simply wind it back up when you are finished and close the cover to secure it away.

Usage
The Polycom Communicator functions very nicely as an external sound device and the quality of the sound on both recording and playback is simply superb. Of course no test is complete without making a bunch of phone calls via a softphone. While several popular softphones where tested, the majority of the calls were placed with Counterpath’s X-Lite.

Besides Skype, our testbed was different trixbox systems running Asterisk Open Source PBX software. One system was connected to over the internet and then dialed out using a PRI circuit. The second was a local machine using VoIP providers and PSTN lines.

Using VOIP providers and PRI circuits the call quality outstanding on both sides and the full duplex operation was flawless. Calling out on a system over a traditional POTS line wasn’t quite as good as was to be expected as the capabilities of the Communicator far surpass the quality of cheap analog lines.

Results
The Polycom Communicator retails for around $149 which is about the same cost as the lower-end Polycom IP phones. This may seem like quite a high price for a simple USB speakerphone but there is nothing cheap or low-end feeling about the device. You get outstanding audio quality on both sides of the conversation as well as being able to use it for audio playback and recording. The integrated USB cable tucks away nicely in the back and means you will always have it available when you need it. Compared with more expensive dedicated speakerphone product, the Polycom Communicator CS100 is an excellent device at a very good price. If you use Skype or another softphone and you need a speakerphone device, you cannot go wrong with the Polycom Communicator. You can also use it as the microphone and plug in a standard cell phone headset into the jack on the side for when you need privacy.

The GXW-4104 is a slick looking box that is certainly one of the best looking products to ever have come out of Grandstream’s engineering department. We wanted to put the GXW-4104 to the test so we decided to put it to use in a production environment in a small business using a trixbox server. First, lets take a look at what’s inside the box.

Features
The GXW-410x is a voice and video gateway that provides a SIP trunk to an IP PBX such as Asterisk, SIPx, OpenPBX, or other compatible systems. The gateway series includes the GXW-4104 (4-port) and GXW-4108 (8-port) models. These units convert a standard POTS (analog) phone line into a VoIP trunk delivered to your IP PBX. The video surveillance port enables remote security monitoring.

• 4 and 8 port media gateways
• Video surveillance port
• Two RJ-45 ports
• Multiple SIP accounts and profiles
• Audio Codecs: G.711, G.723, G.729, GSM, G.726, G.168
• Real time H.264 video codec
• T.38 Fax
• Echo Cancellation
• Dynamic jitter buffer

Setup
My first real gripe with the box is the setup procedure. Every other device I can think of defaults to using DHCP so you can toss it on the network and get up and running as well as use tFTP servers for automated provisioning. Instead, the factory default setting is to have the IP address of the device set to 192.168.0.160 which means you have to tweak your IP address settings to be able to access the unit unless you are already running on that network block.

Once I logged into the web interface the first time I manually set the IP address and related network settings and rebooted. Next we set it up to work with our trixbox platform.

Under the FXO Lines tab you want to change the Stage Method and the Offhook Auto Dial settings. Set the following fields as shown here:

Stage Method (1/2): ch1-4:1;
Offhook Auto Dial (VoIP): ch1-4;xxxxxxxxxx;
(Set xxxxxxxxxx to your phone number for routing within FreePBX)

Select Update and reboot the device. Double check the settings and make sure they are saved, there are a few quirky issues with saving settings sometimes so always double-check to see if it was actually saved.

On the Channels tab there is only one setting you should change here and it is the setting for DTMF type. The default for an Asterisk-based system is RFC2833 so the setting we want is as follows:

DTMF Methods: ch1-4:2;

Again, save your changes and then reboot and like before, double check to make sure the settings are actually saved properly. Hopefully this bug will be fixed in a current firmware version and will work as reliably as other Grandstream products we are used to using.

The final settings are under the Profile 1 tab. Here you want to set the SIP server, registration and NAT options as follows:

SIP Server: set to IP address of your PBX
SIP Registration: No
NAT Traversal: No

For the last time, select Update and reboot and double check all the settings to be sure they are all saved.

All you need to do in FreePBX is to create a SIP trunk. Since we have registration turned off we only need to setup an outgoing trunk so long as you have “allow anonymous sip calls” turned on in the General Settings.

Trunk Name: Grandstream

PEER Details:
context=from-trunk
host=<ip address of PBX>
insecure=port
type=peer

Save your settings and click on the red apply bar.

Under your inbound routes make sure you either have a blank (no cid/no did) route or one with the DID set to the phone number you used during the GXW-410x setup routed to a destination. Then go to Outbound Routes and make sure you have an outbound route setup to use the Grandstream trunk.

Once everything is saved and the red apply bar is clicked, you should be working.

Results
Other than the quirky IP address setup, the reset of the device setup was very simple and fairly straightforward. On my particular line, the receive volume was a bit low so I went back into the settings on the device to crank up the RX Gain a few notches. This resulted in another bout with the wonky firmware saving feature, but a few tries got everything saved and working.

The sound quality is as good as the Linksys SPA-3000/SPA-3102/SPA-400 devices. I have never had echo on my lines with the Linksys so I can’t really say if the GXW-410x devices are any better or worse as I had no echo problems at all.

With a street price of around $279, it is the most affordable 4-port SIP gateway available. I’m not sure how useful the video surveillance option is and it is not well documented. If I can find some video source for it I will give that feature a whirl. In the meantime, the GXW-4104 is an excellent value and performs quite nicely for a unit at this price.

Features
The WBP54G is a realitivly simple device, it is simple a wireless device that connects to a wireless access point and has an ethernet cable that you can plug into any network port to get remote connectivity. While the WBP54G is specifically designed to work with Linksys’s line of IP phones, the fact is, with the correct AC adapter, it can be used with any network device, we will cover the caveats to this later.

• Put your IP Phone wherever you want, with no cabling hassle
• Connects your IP Phone to your Wireless-G network
• Shares power with the IP Phone — only one AC Adapter necessary
• Wireless connection protected by WEP, WPA or WPA2 encryption

When used with the recommended Linksys phones, you simply take the power cable that came with the phone and plug it into the WBP54G and then plug the power connection from the WBP54G into your phone, and plug the network cable into the phone, and you are all set.

Setup
Of course, its not quite that simple to get it working, but its not difficult either. You actually start by powering the device with the phone’s power adapter and then plugging the network cable from the WBP54G into your switch or directly into the network jack on your computer. Launch the CD that came with the device and step right through the setup process.

The setup program will detect any available wireless networks and allow you to connect to them. Once, the setup is complete, disconnect the network cable and plug it into the phone along with the power and you are all set to configure your phone.

As shown here, the WBP54G fits nicely into the phone stand on the Linksys SPA-941/942 phones leaving only the power cable coming out of the back. This makes for a nice clean setup and hides the devices and extra wires.

Usage
When I first discussed this idea with Linksys months ago i had some very distinct purposes in mind, because of the design of our house, it is very difficult to bring cabling into the main living room and the master bedroom so my remote office phone sits inconvieniently on the opposite side of the house. With the WBP54G I now have phones in these locations without having to string any wires. While this works great for me, the possibilities are endless for many offices where cabling is not feasible or desired, or for temporary phone installations where stringing cable would be too costly.

As mentioned earlier, the WBP54G is designed to work with any of the Linksys phones and ATA devices, but what else can it work with? As far as other phones go, the Grandstream GXP-2000 uses the same voltage as the Linksys phones so it is a good fit for that phone. For other phones such as Polycom and Cisco phones, you would need to supply the correct power adapter for the WBP54G and a seperate power adapter for the phone as well. While this may be a little inconvienent, it is still a better bet than wireing in many cases.

Summary
At a retail price of $44.95, the WBP54G is an excellent choice for those situations where cabling just isn’t an option. The device is simple to setup and works perfectly. Kudos to Linksys for coming up with such an affordable option and an innovative product.

Linksys
http://www.linksys.com