Skype over 3G – calculations

With the availability of an iPhone Skype client with background availability, I wanted to find out, how much it would cost me if I move calls from the regular mobile network over to Skype over 3G.

Doing VoIP over 3G is a non-issue from a political standpoint here in Switzerland as there are no unlimited data plans available and the metered plans are expensive enough for the cellphone companies to actually be able to make money with, so there’s no blocking or anything else going on over here.

To see how much data is accumulated, I hold a phone conversation with my girlfriend that lasted exactly 2 minutes in which time, we both talked. Before doing so, I reset the traffic counter on my phone and immediately afterwards I checked.

After two minutes, I sent out 652 KB of data and received 798 KB.

This is equal to around 750 KB/minute (being conservative here and gracefully rounding up).

My subscription comes with 250 MB of data per month. Once that’s used, there’s a flat rate of CHF 5 per day I’m using any data, so I really should not go beyond the 250MB.

As I’m not watching video (or audio) over 3G, my data usage is otherwise quite low – around 50MB.

That leaves 200MB unused.

With 750KB/minute, this equals 4.4 hours of free Skype conversation. Which is something I would never ever reach wich means that at least with skype-enabled people, I can talk for free now.

Well. Could.

While the voice quality in Skype over 3G is simply astounding, the solution unfortunately still isn’t practical due to two issues:

  1. Skype sucks about 20% of battery per hour even when just idling in the background.
  2. Skype IM doesn’t know the concept of locations so all IM sent is replicated to all clients. This means that whenever I type something for a coworker, my phone will make a sound and vibrate.

2) I could work around by quitting Skype when in front of a PC, but 1) really is a killer. Maybe the new iPhone 4 (if I go that route instead of giving Andorid another try) with its bigger battery will be of help.

My favourite asterisk feature

I’ve just included this into the context of the dialplan where the calls from and to our internal phones live.

exten => _55[6-8][1-9],1,SIPAddHeader("Call-Info: sip:asterisk;answer-after=0")
exten => _55[6-8][1-9],2,Dial(SIP/${EXTEN:2})

this is as useless as it is fun.

Too bad softphones are getting some real attention in the office lately, as they don’t support the answer-after feature and even if they did, where is the fun of just making yourself heard on the headphones of the victim as opposed to doing that directy on their speaker – loud enough to be heard in the whole office.

VoIP is fun and it’s about time I do more to our asterisk config but just watch it work and never fail.

More Asterisk stuff

I thought I’d give a little update on what’s going on in my Asterisk installation as some of the stuff might be useful for you:

Speed Dial

If you have Snom Phones and want to program the function keys to dial a certain number, be sure to select “Speed Dial” and not “Destination” when entering the number.

Destination was used in earlier firmwares but it now used to not only make the phone dial that number, but also subscribe to the line to make the LED light up when the line is used.

This obviously makes no sense at all with external numbers and requires some configuration for internal ones (see below). The additional benefit is that buttons with “Speed Dial” assigned don’t turn on the LED.

Dial by click

You can dial a number from the Mac OS X address book aswell. Asterisk will make your phone ring and redirect the call once you pick up (just like AstTapi on Windows). I had the best experience with app_notify. I don’t quite like the way how it notifies clients of incoming calls (hard-coding IP-Addresses of clients is NOT how I want my network to operate), but maybe there will be a better solution later on. Currently, I’m not using this feature.

Dialing works though.

You don’t have to modify manager.conf, btw, if you already have the entry for the AstTapi-Solution. app_notify will ask for username (manager context) and password when it launches the first time.


As noted above, your Snom Phone can be advised to monitor a line. The corresponding LED will blink (asterisk 1.2+) when it’s ringing and light up when the line is busy.

Snom-wise, you’ll have to configure a function key to a “Destination” and enter the extension you like to monitor.

Asterisk-wise you have to make various changes:


  • Add subscribecontext=[context], where context is the context in extensions.conf where the corresponding SNOM phone is configured in. I’ve put this to the [general]-Section because all phones are sharing the same context (internal).
  • Add notifyringing=yes if you have Asterisk >= 1.2 and want to make the LEDs blink when the line is ringing.

This is a bit hacky: In the sip-context add a notify extension for every line you want to be allowed to be monitored. Unfortunately, you can’t use macros or variables here, so it’s messy.

On my configuration it’s:

exten => 61,hint,SIP/61
exten => 62,hint,SIP/62
exten => 63,hint,SIP/63
exten => 64,hint,SIP/64
exten => _6[1-9],1,Dial(SIP/${EXTEN},,tWw)

While I would have preferred

exten => _6[1-9],hint,SIP/${EXTEN}
exten => _6[1-9],1,Dial(SIP/${EXTEN},,tWw)

Though this may have been fixed with 1.2.2, but I’m not sure just yet.

You may have to reboot your phone after making the configuration change there. To check the registration in asterisk use SIP show subscriptions.

You should get something like this:

asterisk*CLI> SIP show subscriptions
Peer User Call ID Extension Last state Type 62 3c26700b57e 61 Idle dialog-info+xml
1 active SIP subscription

This is not quite tested as of yet because the guy at extension 61 is currently in his office and I don’t want to bother him ;-)

Update while editing/correcting this text: It works. They guy has left and I checked it.

Asterisk Extended

Playing around with Asterisk, it was inevitable for me to stumble upon AGI.

AGI is a protocol quite like CGI which allows third party applications to be plugged into asterisk, giving them full control over the call handling. That way, even non-asterisk-developers are able to write interesting telephony applications.

One thing I always wanted to do is to set the CallerID on incoming calls. Some numbers are stored in our customer database. There is no reason not to show the customer names on the phones displays instead of only the number.

The snom phones do have a little addressbook, but it’s very limited in both amount of memory and featureset, so it was clear that I’ll have to set the CallerID via Asterisk (SIP allows for transmission of a caller-id. And so does AGI)

Additionally, I thought, it would be very nice to use the swiss phone book at or even the non-free ETV to try and guess numbers not already in our database.

That scenario is exactly what AGI is for.

As AGI works like CGI, it creates a new process for every call to AGI applications. This is not an option if you want to use interpreted languages. Well. it *is* an option considering our low amount of calls we are getting per time unit, but still. I don’t like to deploy solutions with obvious drawbacks.

Besides, launching a PHP interpreter (I’d have written this in PHP) can easily take a second or so – not acceptable if you want the AGI script to be mandatory on each call. Think of it. You don’t want the caller to wait for your application.

The solution to this is FastAGI, which works like FastCGI: A server keeps running and answers AGI-requests. Like this, you start the interpreter once and just serve the calls in the future. You save the startup-time of the interpreter.

Even better: It allows to run the AGI applications on a different machine than the PBX. This is good because you want the PBX to have as much CPU time slices as possible.

Unfortunately, this made PHP quite unsuitable for me: While it is possible to write a socket server in PHP (ext/posix does exist), I never managed to get it to work as I wanted to. It was slow, unstable and created zombies.

Then I found RAGI which was even better. For quite some time now, I have been looking for an excuse to do something with Ruby on Rails. With RAGI, I finally got it.

Getting the sample provided with RAGI to work was very easy (look at the README file). And reading through that sample file, I was very pleased to see the simplicity of writing a AGI-Application in Ruby (RAGI uses FastAGI, of course).

Now I can finally start hacking away in Rails to create my internal-customer-database / external-phone-lookup application (with some nice caching/timeout handling) to finally show the name behind the calling phone number on the displays of our SNOM phones.

Of course I’m going to provide the sourcecode here once I’m done.

Snom 190

The Snom 190 is a SIP hardware phone which I have ordered recently to continue my asterisk experiment.

Yesterday it arrrived.

I have to say: I love that device. Contrary to those proprietary PBX phones, the Snom 190 is easy to use, provides a big heap of features (complete remote managability, web interface, dialing over http-request (outlook-plugin – here I come)) and does not cost more than what the other companies ask for their lowest entry level phones. The Snom even looks good!

Like many other devices today, the Snom 190 runs Linux (2.4), though this time I have not tried to hack it yet. All the sources including the developement environement are available at the website of snom.

Contrary to the somewhat crappy ZyXel 2000W which I have tested too, the Snom 190 is ready for productive business.

This makes implementing VoIP at our company seem more and more likely every day.

Asterisk – it’s getting real

Last week I talked about me and Christoph installing Asterisk on my thinkpad to do a little VoIP-Experiment.

While we were able to create a should-be-working configuration, actually calling to the outside PSTN network did not work. Read the details in my other article.

Last saturday, we fixed that.

There seems to be a problem somehwere between the AVM CAPI Driver and the CAPI layer of the 2.6.11 kernel. After we downgraded to 2.6.10, the problem solved itself without we doing anything more.

So… this was getting interesting…

The first thing I did was to annoy my wonderful girlfriend:

exten => s,1,Wait,1 ; Wait a second, just for fun
exten => s,2,Answer
exten => s,3,MP3Player(/home/pilif/mp3/3.mp3)

(included into or used as the default context)

Where 3.mp3 is that endlessly stupid song “Tell me” (or whatever it’s called) by britney spears (this is an insider-joke – both of us just hate that song). Then I told her to call that number…

While this example is completely pointless, it was fun to watch my girlfriend connecting and listening to the song (which soon ended in a disconnection log entry)

exten => s,1,Wait,1
exten => s,2,Dial(SIP/12345,60,tr)
exten => s,3,Congestion

This makes much more sense and directs all incoming calls to the SIP-Phone 12345 as configured in sip.conf. After 60 seconds, it sends back a congestion signal. The first entry would not be necessary, but I hate it when I call somewhere and the phone is answered just at the first ring. So in my PBX, the answering party will wait one second before directing to the sip-phone.

In musiconhold.conf I’ve configured madplay as my MP3-Player for music on hold:

default => custom:/home/pilif/mp3/,/usr/bin/madplay --mono -R 8000 --output=raw:-

madplay is much better than mpg123 used per default as it accepts VBR encoded input and bitrates > 128 kbit which is what nearly all of my MP3’s are encoded with.

In zapata.conf enable music on hold with musiconhold=default in [channels]

The next thing was an optimization of the SIP-Phone used…

X-Lite is nice, but in the end it’s just a demo for other products by the same vendor. Call transferring is not possible for example, which is what we wanted to try next.

The best soft phone we’ve seen so far is SJPhone. A configuration guide is here

But the real clou is the Zyxel 2000W phone that’s currently on my desk: The phone has a WLAN interface (unfortunately no WPA support) and can perfectly well speak with asterisk.

The phone has some problems though: it’s slow, it has no support for call transferring, nor holding, neraly every configuration change causes it to reboot,… In the end I really hope Zyxel will further improve the firmware, which is what they seem to be doing – the current release is from the end of february, so quite current.

The next thing will be trying to install a webbased frontend to asterisk and creating a real dialplan with voice mail. Then, our experiment will be over and we’ll see how it can be put into practical use (like finally getting rid of the old, proprietary PBX from alcatel of our landlords)

Fun with VoIP

When I read for then n-th time about Asterisk, an Open Source PBX solution, I deceided to team up with Christoph and tame the beast.

I have actually two problems with asterisk as it stands now:

  1. There’s not much really useful newbie-documentation or tutorials. There are some sample configurations, but they are not very useful because…
  2. the tool has a incredibly intransparent and difficult to understand syntax for it’s main configuration file (extension.conf). I’t just like it’s with sendmail: Many extremely low-level things to care of for getting complex high-level results.

I thought, that teamed up with Christoph, we’ll more likely to see some results.

The first thing was defining the parameters of our experiment. Here’s what we wanted to do:

  • Act as a SIP-Proxy, so two softphones (we did not want to buy too much actual hardware yet) could talk to each other.
  • Provide a gateway to the ISDN-Network, so both SIP-Phones can dial out to the rest of the world.
  • The same gateway should be able to receive incoming calls and direct them to one of the Phones (just one for now).

In the next session, we want more advanced features, like voicemail and waiting music. A third session should provide us with a webbased frontend (I know there are some). But for now, we wanted to concentrate on the basics.

The next step was to get the required hardware. I already have Gentoo running on my Thinkpad, so that was a good base. Furthermore, we needed any ISDN-Solution being supported by Asterisk. As we had a plain old BRI interface and a very limited budget (it was just an experiment after all), we went with the Fritz Card USB by AVM which has Linux CAPI drivers, albeit only binary ones (we could also have used the PCMCIA-version, but this is three times as expensive as the USB one).

Said piece of hardware proved to be a real pearl: It’s very compact, does not need a power adaptor and was very easily installed under Linux. I would not be using this for a real-world solution (which most likely requires PRI support and absolutely would require open sourced drivers), but for our test, this was very, very nice.

Installing the needed software is where gentoo really shined as everything needed was already in the distribution: After hooking up all the stuff, we emerged net-dialup/fritzcapi, net-misc/asterisk and net-misc/asterisk-chan_capi which suked in some more dependencies.

The next step is to reconfigure the kernel for the CAPI-stuff to work. Just include everything you find under “Device Drivers / ISDN Support / CAPI” – even the one option marked as Experimental (as the CAPIFS is needed and only available when enabling “CAPI2.0 Middleware support”)

Then, we made sure that CAPI (a common ISDN access API) was running by issuing capiinit start.

Then we went on to asterisk.

The fist thing, you have to do is to set up the phones you’re using. As we worked with SIP-Phones, we used sip.conf:

port = 5060
bindaddr =
tos = none
realm =
srvlookup = yes

context = theflintstones
dtmfmode = rfc2833
disallow = all
allow = gsm
callerid = "Fred Flintstone" <12345>
secret = blah
auth = md5
host = dynamic
reinvite = no
canreinvite = no
nat = no
qualify = 1000
type = friend

accountcode = 12346
dmtfmode = rfc2833
host = dynamic
auth = md5
secret = blah
canreinvite = no
context = theflintstones
qualify = 2000
type = friend
disallow = all
allow = gsm

This worked with our two test-phones running X-Lite

Interesting are the following settings:

realm The realm. I used our internal domain here. The default is asterisk. Your VoIP-Address will be identifier@[realm].
accountcode This is the username you’re going to use on the phone
context The context will be used when we create the dial plan in the feared extension.conf

Then, we configured CAPI in capi.conf



Those settings are said to work in Switzerland. Interesting is the setting for msn. This is where you enter the MSNs (phone numbers) assigned to your NT. I somewhat X-ed it out. Just don’t use any leading zeroes in most countries. You can enter up to five using commas as separator.

The next thing is to update modules.conf. In the [modules]-Section, add load =>, in the [global]-section, add

Without those entries, asterisk will complain about unresolved symbols when loading the CAPI modules and will finally terminate with a “broken pipe”-Error. Thrust us. We tried. ;-)

The best thing now is that you can already test your setup so far. Launch asterisk with asterisk -vvvvvc (each v adds a bit of verbosity, while -c tells it not to detach from the console). If it works well, you’ll end up at a console. If not, make sure, that capiinit did not report any error and that you’ve really added those lines to module.conf.

Now for the fun of it, call one of your MSNs with any phone.

Asterisk should answer and provide you with a demo-menu

The next step is configuring extensions.conf. This is somewhat complex and I will go into more detail, as soon as I’ve figured out, what’s wrong with our test-configuration. We’ve added this to the end:

exten => _0[1-9].,1,Dial(CAPI/44260XXXX:b${EXTEN},30)
exten => _0[1-9].,2,Hangup

include => ch-fest-netz

Just look that you enter one of the MSNs you have configured in capi.conf.

Now what this configuration should do is to allow those SIP-phones (recognize the “context” we used in sip.conf?) to dial out via CAPI.

You best learn how to configure this beast by calling the demo-voicebox and then comparing the log output of Asterisk with the entries in extension.conf. Basically, exten => defines a dial plan to execute. Then comes the pattern of numbers dialed to recognize. After that comes a (BASIC-like) sequence-number, followed by the action to execute.

The format of the number-pattern is explained in one of the comments in extension.conf

Now, this configuration does not work for us: When I dial on the SIP-Phone, Asterisk notices this, actually connects the ISDN-line (the target phone actually rings), but does not seem to notice when the target phone is answered.

If I answer the phone, it’s just silence in the line. The SIP-phone is still in the “trying to connect”-state.

This stays this way until I cancel the dial attempt in the SIP-phone. After that, asterisk prints more log entries – one of them the notice that the connection was successfully established.

A question in the malinglist was promptly answered: My configuration is correct, but maybe I’m running into a bug of Kernel 2.6.11. I was told to downgrade to 2.6.10, which is what I’m going to do next.

After this, I will extend the dial plan so I can call the internal SIP-phones both from another softphone or from a real phone over the ISDN

It’s hacky, it’s just somewhat working, but it’s a lot of fun!

I’ll keep you updated.