After getting VoIP phone numbers at a SIP provider for our Podcast KUNZ and a couple of students as well as myself I had to try out a new gadget that hit the markets last month. It’s a phone by Siemens that offers PSTN (i.e. the normal phone line) as well as VoIP services on six different VoIP accounts that are freely configurable.
Problems I had
- Trying to configure the base via web browser turned out to be a problem, since from my server’s perspective the base is in the external zone, thus I’d access it via a Squid2 proxy. That just does not work as I keep getting incomplete HTML pages. Why? Beats me! Other hardware such as the router can be accessed without any problems. So I had to get a direct connection between browser and base.
- Then I tried to connect my old DECT phones (i.e. handsets) to this new base. It worked fine for other Siemens and Panasonic models, however, my good old Audioline phone just won’t get recognized by the base. I don’t think it’s the handset as it has connected to other bases without any problems.
Things I like
- Other than that the configuration was pretty straight forward. The web interface lets you set up LAN as well as six SIP accounts for VoIP plus something called gigaset.net which I disabled immediately not knowing what kind of data gets transfered to it.
- The color display of the handset is nice, though a little small by today’s standards.
- It’s possible to backup the handsets’ phone books via the web interface into a .tsv file. It would be more useful, though, if there was a script to convert a .vcf file into a valid .tsv for this phone.
- It supports STUN.
Things that should be improved
Maybe Siemens will make use of the firmware upgrade feature to enhance this product in the future, because there are some aspects that just don’t seem right.
- Having six SIP accounts for inbound calling is great. But you can only use one for outbound calling per handset. So in essence you need to use six different handsets to use all six SIP lines. Now here is a suggestion on how to improve this: The phone distinguishes between PSTN and VoIP calls by an * as suffix (i.e. 1234 calls via PSTN, whereas 1234* places a call via VoIP). They should use a similar suffix (or prefix) to choose between the various VoIP accounts. That would make things a lot easier.
- It would be nice to be able to switch VoIP on and off via the handset as I don’t need the router to be online all night.
- URI dialing would be nice. This way you could call URIs directly such as 613@some-other-sip-provider.com, thereby saving a lot of money. Otherwise you are stuck with whatever your SIP provider has to offer.
- The date and the time are entered through the handset. What about NTP? A device hooked up to the internet should be able to get the exact time automatically.
- The aforementioned proxy trouble needs to be resolved.
Cosmetics
- The ringback tone generated by the base when calling a VoIP number is just horrible. It almost sounds like a German ringback tone but it is much faster. It makes it difficult to tell the difference whether it’s a busy or a ringback tone – actually it sounds as if the phone was kind of nervous.
I’d like to switch the tones to the American standard.
Linux – Getting the penguin into it?
I noticed that the phone relies on open source software. So I wonder if it will be possible in the future to use an alternative firmware for the base as they do with so many cool gadgets nowadays (like the Linksys router or the NSLU2 etc.)?
Summary
It’s a nice gadget and I am looking forward to gather more experience while using it in the future. However, the first impression already reveals that it is not perfect. Hopefully Siemens will improve this product via firmware upgrades or offer open source programmers access to their firmware.
[Update]
With a free firmware update the guys at Siemens added the optional suffix dialing mentioned above enabling you to choose various providers on outbound calls. So, Kudos to them!
