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Archive for February, 2009

Fritz Box fon WLAN 7050

Tuesday, February 3rd, 2009

After I plugged the Fritz! box in for the first time and configured it to use Voipstunt for all international calls, I reached for my ISDN phone connected to it and placed an international call. I was surprised by the little message appearing on ISDN phone display saying “VOIP 1723″. Wow! That was neat. For curiosity sake, I tried a sample toll-free number which was meant to go through PSTN. The phone now said “ISDN 328033″. Cool.

It seems that Fritz! is loaded with many such gems such as the one described above.

The 7050 (and its cousins 5050 and 7170) are one of the few (or maybe the only) affordable ATAs with ISDN BRI ports for both FXS and FXO interfaces. That means it can act as an ISDN PBX for up to 8 ISDN phones hooked to the FON S0 port on the Fritz, and yet you can add 3 analogue extensions into picture. The external PSTN line (FXO) can be either analogue or ISDN.

The web configuration interface is very well laid out and it’s geared towards simplicity. It has two modes: standard and expert. As expected, the expert interface gives more configuration option and is necessity, for example, if your Fritz is behind a cable modem.

VoIP accounts configuration

Fritz! supports up to 10 VoIP accounts and can be registered on all of them. International prefix and country code, as well as local area prefix and code can be configured separately for each account, so that Fritz! can do a simple number manipulation.

For example, Voipstunt requires all calls to be placed as 00 + country code + area code + subscriber number. Now, if your Fritz! sits in some London office and you want to avoid dialing 00442012345678 every time you call home from work telling you’re late, you can configure ‘00′ as intl. prefix, ‘44′ for country code, ‘0′ for area prefix and ‘20′ for area code. In that configuration, all numbers that have neither intl. or area prefix will be considered local, and Fritz! will prepend 004420 in front of them. If a number starts with single digit 0, it will be stripped and 0044 prefixed to it. If a number starts with 00 (intl.prefix) it will be dialled as is.

You can configure each analogue port to react to incoming VoIP calls for one or more VoIP accounts as well as ISDN MSNs.

To chose which VoIP calls you’ll be receiving on your ISDN phone, you have to configure it on the phone itself. This might be a bit problematic since most phones have limited number of configurable MSN entries, and that limit can be very low. Bear in mind that each VoIP provider is allocated an arbitrary internal (virtual) MSN, that must have a matching MSN entry in your phone if you want your phone to receieve incoming calls for that VoIP account. Or, you can delete all MSN entries in your phone, so it will react on all incoming calls.

That is really nice ISDN trick since you can find out, just by looking on your phone display, which VoIP account is ringing. “Call to 1723 from xxxxx” or similar (depending on your phone) indicates that you just received a call on your Voipstunt VoIP-in number.

So to enjoy full functionality of your Fritz! 7050, an investment in ISDN phones might be very smart decision, regardless of the type of your PSTN connection.

The dialing Rules

I was a bit disappointed at first to realize that no complex dialing rules are supported. Fritz! looks for the longest entry in the dialing rules that fully matches from the start of the dialled number and that’s it. So given a sample dialing plan:

00: Voipstunt
003859: Voipfone

… calls to Zagreb (003851xxxx) will go over the Voipstunt account (the 003859 dialing rule doesn’t fully match), while calls to Croatia mobiles will go over Voipfone.

Call by Call

In some countries it is possible to select a PSTN operator by placing an appropriate prefix. You can define up to 10 prefixes that are triggerred by the dialing rules.

Call Through

Basically, you select the MSN for which you want to enable the Call-Through via a drop-down select. Keep in mind that MSN can be either PSTN numbers or VoIP virtual MSNs.

When Fritz! box receives the call on the MSN designated for Call-Through, it answers the call with a special tone, awaiting for caller to supply the PIN (4 digits). If PIN is correct, Fritz! sends a short beep as a confirmation. At this point you’re hooked to Fritz! PBX, so that you can enter local extensions to reach the phones connected to your Fritz! or you can dial ‘0′ for outside line, and then the desired number.

The dialing is then performed using the defined dialing rules. If none matches, the dialing is performed using the preferred method defined for Call-Through (ISDN or selected VoIP provider). When a call to final destination has been established, it can be terminated with a pound key (#). You’re now back to PBX and you can place another call (don’t forget 0 for external line) etc.

The access to Call-Through functionality can optionally be restricted by Caller ID. The access list holds max. 8 entries.


If you want to use 7050 as a DSL modem, you connect the output from a splitter to the DSL port. In that case DSL connection can be shared by computers attached to both LAN A and LAN B ports.

In case you have a cable connection (or any other broadband connectivity that ends up with good old Ethernet), you connect it to LAN A port and select the appropriate option in the ‘Internet’ configuration page. You can further select whether the Fritz! will provide a NAT routing to the computers attached to the LAN B port.

Back Panel

FON SO is used to connect your ISDN terminal equipment.

FON 1, 2 and 3 are for analogue telephones.

USB port can only be used to provide network connectivity to a computer attached to it.

LAN B and LAN A ports are described above.

POWER is 12V 1.2A DC.

Wireless can be set up in normal (AP), repeater (WDS) mode or completely turned off.

Front Panel

Only 5 LEDs are provided, which somewhat hides the functionality and built-in complexity.


“Internet” with a handset icon. Lit when VoIP call is conducted.

“Festnetz” with a handset icon. PSTN call is underway.

“DSL”. Probably lits when DSL signal is present (couldn’t check it
b/c I have cable). For cable users, lits when DHCP lease is obtained or PPPoE session established (depending on the encapsulation used by your ISP).

“INFO”. Blinks during firmware upgrade.


In summary, a combined DSL modem and wireless router, ATA and PBX. Unfortunately it doesn’t make coffee!

Positive: Excellent price/functionality ratio

Negatives: does not support G729 or G723 and no support for in-band DTMF

In final conclusion, the Fritz! is an excellent device and I would recommend it. However, lack of in-band DTMF and support for G723 and G729 codecs lowers the final rating to 4 1/2.

This review is based on Fritz! box fon WLAN 7050 (or just 7050) available across the whole of Europe. It’s not, however, the same device that Sipgate sell. The Sipgate device has the same overall functionality, but lacks the LAN router and WiFi AP elements.

DU@LPHONE USB Cordless Skype Phone Review

Tuesday, February 3rd, 2009

Being a cordless telephone user for years now, I was curious as to how the Skype DualPhone would work and its future sister SIP based product which due out soon.


You get a good feeling when opening the box – all components are nicely wrapped including: a multi-lingual manual, a little cheat card, driver CD, the telephone with its battery, base station, power supply and a set of different connectors for the different telecoms outlets in various countries.


The first thing to do, as with any cordless device is to take out the base station, plug in the power supply and put the battery in the telephone so that it can do it’s first 24 hour charge as stated in the manual. The following day I took it to the office for a full test, first plugging the USB cable to my laptop before running the install procedure for the “telephone suite” on the supplied CD. When running the installer you can either download the latest version via the internet or use the CD. One smooth install later, and after allowing the telephone suite access to my Skype client via the pop-up that skype gave me, I was up and running.

Getting and making calls

The first test I made was to add “echo123″ to my contact list in Skype (the Skype echo test service). Then on the telephone pushing the green horn button allows you to scroll through your online contact list and select the person you wanted to call (the telephone suite allows you to hide and un-hide different status’ like Offline, Do not Disturb, Away etc. from your telephone’s contact list.

All sounded nice and clear on the echo test when wearing a headset. Since initial testing with the Skype echo server, I have made several calls, some of over an hour long, which all have worked very well. I also tested my SkypeOut account which, as to be expected, worked just fine.

Now comes the handy feature of this telephone and that is that you can plug it in to your exsisting landline turning it into a combined PSTN/Skype telephone, explaining the name ‘DualPhone’. Although the audio on the PSTN calls wasn’t as high quality as my normal cordless, it was more than sufficient (bear in mind my normal cordless telephone had cost double the price of the DualPhone).

Skype Features

The telephone, once connected via the PC’s USB port, hooks up to the Skype API which allows you change your online status via a menu. Also any status change on your contact list generates a beep and lights up the display with a notice – handy to keep track of who’s online or when one of your buddies comes on.


Certainly as good as any other cordless I’ve used so far – all around my home/office, in the garden and even when going a bit further afield it all works as expected.

Battery Life

Battery for me tends to run at about 3 days, which is OK for a cordless – at least up to scratch with other such telephones.


I really like this telephone, have been using it now almost on a daily basis and it’s perfect for both for my wife and I (which can’t be said for all my geeky toys). I give the telephone a 9 out of 10. It didn’t quite make a 10 as I think it misses 3 features:

1. Not able to check your SkypeOut credit via the telephone

2. Not easy to understand why a call failed, e.g. wrong number or because there is no SkypeOut credit left?

3. A filter whereby buddies can be announced on the telephone, or maybe which buddies shouldn’t be.

Apart from those 3 items, it’s a great telephone and well worth the money. Can’t wait until they bring out their SIP/PSTN dual cordless.

Our thanks to Du@lphone for the review sample.


Sunday, February 1st, 2009

I was struck by the low price of this phone and managed to get one for testing.

This is a two-line IP phone and can register to two independent ITSP’s (Internet Telephony Service Providers). The two lines can be switched by pressing a key and the display indicates which line is currently in use.

In addition it supports 3-way conference and works well with Asterisk (although it’s SIP only). I setup a 3-way conference on an Asterisk platform facility, making two calls provided by two separate ITSP’s.

Finally, the most interesting thing I found from this phone is that it could show me specific call status indicators such as SIP 100 trying and 180 ringing etc on the LCD. It may be of no use to your average customer, but it is very useful when making a system deployment.

You can get the specifications for this Phone from the company website here:-

In Use

The phone in this review is the black model without PoE (Power over Ethernet). I like the appearance and design of the GIP300 – it would comfortably suit either office or home environment and can be wall mounted. It features a 10/100M Ethernet port, one 5V power adapter jack (just a little bigger than my thumb) and one 3.5mm headset jack.

It was easy to understand and operate the keypad even without the manual. The two Ethernet ports can be configured as either switched or router mode. This design is quite helpful from a security perspective. When it is configured as switched mode it simply acts as a two port switch. When configured in router mode, it acts as a NAT device with PPPOE and DHCP Server functions (a SoHo router in other words).

The LCD display is abundant with details – the status of the two Ethernet ports, the history of missed calls, call transfer, received/dialed call and message waiting indicator all shown clearly. There is also a green LED on the phone panel to indicate registration status.

Besides the IP address assignment, the more advanced settings can be configured through a simple web-based configuration page. There is a curious item entitled “VPN” which appears to be a special function for use with another of the same model phone, enabling P2P interconnection over VPN and avoiding ports blocking of VoIP by some ISP’s. [Sounds interesting - if anyone has any info on this, please send it in... Ed]

The GIP300 supports NAT traversal completely. I have tested the setting of the Outbound Proxy which can cooperate with our system very well. I have not tested the device with STUN.

The GIP300 can support downloading files directly including firmware, ring tongs and dial plan. All downloaded files can be selected from the web-based configuration page. Settings are retained during firmware updates.

The GIP300 supports entire dial plan settings, something I have only previously experienced with Sipura products.


First, it’s a pity that the LCD display is not pixel-based but a segment digital screen. Secondly, the 8 memory keys (M1-M8) are also used for other functions, making the front panel a little complex.

Finally there is no IAX2 support.


All in all a great product at an affordable price. It’s 2-line ITSP facility, 8 memory key and 3-way conference functions collectively make it a good business phone. Coupled to the low price, it turned out to be the IP phone that I’m looking for.

Avaya Callmaster V

Sunday, February 1st, 2009

The Avaya Callmaster V is a state of the art phone unit. Or at least it was state of the art when it was launched, which was probably around the time I was born. So lets review this device.

Looks – In terms of how the Avaya Callmaster V looks, well… its erm…. Not very interesting. The colour of the casing is an uninspiring dirty grey colour, not crisp enoung to be silver and not dark enough to be black (either of which would have been better). The buttons on the device are coloured a lighter shade of grey (not so dirty as the casing). As you can imagine, the grey on grey colouring really blows your mind. It doesn’t really. Its dull as dishwater as my mother used to say, or at least what she would have said had she ever come across the Avaya Callmaster V . In terms of visual excitement there is only one highlight – the hold button. The hold button is a vibrant red colour, which in this sea of grey blandness is the equivilant of a glass of ice water after crossing the desert. This button titillates and excites your senses after the deprivation of the casing and button combo.

There is a screen on the Avaya Callmaster V, but it is really not worth getting excited about. It is a little LED strip big enough for two layers of text. It gives some basic functionality to some of the control buttons, but it is not anything worth getting worked up about.

The phone has the capability of handling two phone lines, originally labelled as line a and line b.

You have a loudspeaker option for when that handset of yours gets too heavy, or if more than you wants to listen to whats being said. The speaker on this option is pretty powerful and so works well, it is slightly restricting because of its ability to pick up voices however, so you are not able to walk around the office and have a conversation.

There is a mute button, which I have found to be great fun for making fun of people when they are on the phone. They can be talking away and you can mute them so that they cannot hear you just long enough to swear at them or call them a name. Its amazing the things that you will do for fun when you are bored.

You have the option of transferring calls to other lines, which is a little bit of genius. You can talk to someone and when you get bored of them you can place them on hold, dial up someone that you don’t like and the transfer the person who bores you through to the person you don’t like. I use the Avaya Callmaster V at work but I really want one in my house too.

The next option is for merging calls. This way you can have three people on the line at any one time. This actually has a proper business use, as conference calls can be really handy, or even outside of work you can conference in both your parents if you want to chat to them. I use it for my brother and sister, so that I can call them to arrange birthday parties and such for my parents.

A great device, but ugly.