Welcome to my world

Change SVN repository location

For some reason, repositories can be moved around. If you are using a working copy to develop a program or web-application or whatever, the move usually only implies a 'check-out' and you are set.

However, sometimes the repository is part of a production environment. To be precise, the production version is prepped in a special branch and then deployed using Subversion. Upgrading a website to a newer version only requires a 'svn up' and if you automate the database upgrades as well, web management becomes a breeze.

You might imagine that moving the repository in such an environment is less than desirable: each site or application which is part of that repository becomes un-upgradable and checking out everything by hand and setting it all up (plus the verification that every part is still functional) becomes a time consuming task.

However, the SVN information is all stored in plain text files. The following snippet switches the repository from to svn:// This means you can quickly switch protocol, host names or even the location within a certain host. Simply copy-paste the code below into a file in the directory holding one or more projects for that repository, change the URLs (do not forget to escape by replacing '/' with '\/') and run the script.

DIRS=`find . -name ".svn"`
for dir in ${DIRS}

echo "Processing $dir"

FILES=`find $dir`

for file in ${FILES}


if [ -f ${file} ]; then

echo "Fixing '${file}'..."

sed 's/http:\/\/\/\/' "${file}" > "${file}.tmp"

mv "${file}.tmp" "${file}"




Radio silence

Visitors dwindling to my site, I decided it was time to write something again after 3 months. Its not that I stopped working with computers but rather that my attention moved from working with computers to studying Computer Science. Something that is hard to combine if you are making a dash to the finish line.

In other words: I am finishing my last courses in the coming days and will begin on my thesis. I haven't decided yet if I will post things from my thesis yet (as extra content). Not so much because of the extra time but also because not everything is interesting and some information is classified - making other things inexplicable or incomplete if I were to post it online.

On to more important things: KDE 4.4 is about to hit the virtual shelves! Besides more polishing and some other features, it finally comes with a full blogging client (as part of the base system).

"What", you say, "they fixed KBlogger?" Heck no, KBlogger is still dead as a door nail and as buggy as it has been forever. It seems that someone got the bright idea to actually replace KBlogger altogether with Bilbo - the tool I talked about before. One of the major issues Bilbo had, was that fact that nobody knew it. Personally, I think this comes from the name: Bilbo - as it is also the name from one of Tolkiens books. If you now think "Lord of the Rings", you should be ashamed: Bilbo is part of the story *before* LOTR, he just happens to be alive in the epic story afterwards...

But I digress.

The fact remains that finding the program (even if you know that it exists) becomes hard because all search engines will refer you to hits about the wrong topic. Also, as you might imagine, some folks (*cough*copyright holders*cough*) might have issues with the naming. So it was renamed from Bilbo to Blogilo a while back and as such can now be part of the whole KDE experience. Hurray!

Next time I will probably explain about a project (formal model checker, based on the Promela language) I have been working on and the surprisingly bad hash functions I found on the internet...

Fixing Vmware mouse flicker

After upgrading a VM from Ubuntu 9.04 to Xubuntu 9.10 I noticed that the mouse cursor works fine in a certain region (say 640x480) but outside that region, it flickers between the actual mouse position and the edge of the region.

After reinstalling X a few times and upgrading the VMware tools I found the solution: it is a bug in the newer GTK versions. Since I amd running Gentoo (unstable aka bleeding edge) I am one of the lucky owners of this new bug.

The solution seems to be to instruct VMWare Player or Workstation (I use 6.5.x) to use its bundled GTK instead of the system one:

echo "VMWARE_USE_SHIPPED_GTK=yes" >> /etc/vmware/bootstrap

VoIP providers

In my quest for VoIP support in the Netherlands I found a lot of providers. Most of them seem to be catering the consumer needs rather than the prosumer or even business needs. Most providers fall under one of 3 categories: consumer (“free” landline calls but usually not unlimited call time resulting in paid minutes after a certain time, ), prosumer (better defined call rates, subscriptions, professional level support) and corporate (meant for large businesses, lots of trunks, hosted PBX and so on).

Note that I omit the service providers that only provide services for their own customers. For example, Tele2 has a business oriented VoIP division but also provides VoIP telephony to their DSL customers.

Known ISPs with a VoIP service for their customers: KPN, Tele2, XS4all, @Home.

Note: the listing above is incomplete and based upon results I found in my search online.


VoIP Goedkoop

This is a handy site containing links to multiple “free” VoIP providers. These are fine for consumer usage, normally are not suited for business usage (free equals no quality guarantee, no subscriptions) and usually do not come with DID services.


As one of the more wide known service providers, VoIPBuster advertises with free calls. The “free” here is relative as not all destinations are actually free and for example land line calling in the Netherlands is “free” for a minute (they call that a “Superdeal”, a choice of words which is not reflected by their meaning), after which an unspecified fee is charged. Also, every 120 days, a user needs to reactivate their account by paying 10 Euros, which is converted into 10 Euros credit. Also, mobile numbers cost 25 cpm, which is twice as much as every other VoIP provider I found, not a bargain at all…



VoipQ supports subscriptions, clear call rates and supports DID services as well.


SpeakUp also provides subscriptions, DIDs and clear calling rates. The also offer hosted PBX or just trunk services.

Voys Telecom

Supports VoIP subscriptions from one to multiple lines and everything that comes with it. Does not seem to be the cheapest provider but they seem to have a good reputation.



KPN is the leading Dutch telecom company for decades. Slowly they start to move from POTS solutions to VoIP, for example for their DSL and fibre customers. For the corporate markt they only aim for the big players, as only the start fee is 2000 Euros at the time of writing and subscriptions are expressed in 3 digit monthly fees for the smallest packages. Great for huge corporations and call centers, but useless for everyone else.

Asterisk PBX ramblings

As I am playing with AsteriskNow in a virtual machine to see how I can extend our current PBX (old, analogue, missing lots of options and has zero flexibility) to use the power of a virtual PBX and all its options.

First off I want to explain some terms, you will find these plastered over guides and they confused me as first. This is what I distilled from all that.

VoIP – Voice over IP, internet telephony. Everyone is talking about it and while the name is accurate (you send voice or phone data over a switched IP network, the internet or LAN), it does not describe how this is done. Most people assume VoIP is a technology on its own, it is not.
SIP – SIP stands for Session Initiation Protocol. This is an actual implementation of VoIP. Although SIP has a lot of advanced features (deterring new users to dive in and understanding how it works, like it did me), most people will only use a very, very small subset. See below for more information.
FXO - Foreign eXchange Office, in a nutshell this is an adapter to hook up to a phone line or PTSN (it acts as a client on a line).
FXS – Foreign eXchange Station, this is an (internal) line used to run to a phone or fax machine. The FXS port powers the line, enabling phones to function as well as provide the dial tone and ringing features.
PTSN - Public Switched Telephone Network, this means a phone line. Either a line to a PBX or to the outside world. Also called a trunk.
POTS - Plain Old Telephone System, the old analogue phone system as we know it. The counterpart of this is of course VoIP.
Trunk - A trunk is a phone line, usually only used to denote lines to the outside world. These days a SIP account allowing you to call to the outside world is called a trunk (as it has the same usage as the old physical lines that were connected to a PBX).
DID - Direct Inward Dialling usually denotes the service that connects the POTS telephony system to the digital world of the internet. In other words, it allows you to have a normal phone number which is converted into a VoIP number, which in turn can be hooked up to a VoIP phone or PBX. Usually this is nothing more than a SIP account at a service provider, making usage pretty simple.
Presence – SIP Presence is a technique where a phone tells the PBX it would like to monitor a certain extension or trunk. Normally this is used to light up LEDs on the phone if an extension or trunk is busy. Asterisk has full support for this but most entry-level phones do not sport this feature (more expensive phones targeted for corporate usage have this).

The biggest downside to the VOIP hype is the confusion that it creates in its wake. Like I said, everyone has heard of the phrase internet telephony and most know the term VoIP. Most people there for buy products with big labels on them, marking them as “VoIP phones” and such. The problem is that not always all products work together.

The upside is that most systems fully support SIP which in turn is pretty much a global standard. SIP works by connecting a client to a server, where the client is usually the phone and the server usually is a PBX. The phones are full computer systems on their own, working on the LAN most offices already use. After a phone registered itself with the server using its name and password (a SIP account), the server can reach the phone and vice versa when needed.

When the user picks up the horn and starts dialling, the phone generates a fake dial tone (there is no dial tone as this is a digital system) and waits for the user to start pressing buttons. Every status change (pick up, press number etc) is passed to the server/PBX and is processed to determine who the user is attempting to call. The PBX will then forward the call to another extension or to a trunk and the sound data is transmitted between caller and callee as soon as the connection is complete.

The reverse also holds. After the extension is registered at the server, the PBX can let it ring and after the user picks up the horn, the audio connection is made in order to complete the calling sequence.

So far, this is a simple client-server setup and the only choices one has to make is to determine which audio compression is chosen. The rule of thumb here is that better audio quality comes at a higher bandwidth cost. Most PBX-es like Asterisk convert audio codecs on the fly if for example the remote SIP trunk not supports a certain encoding.

Finally there is the option to call directly to a SIP server or PBX. For example, you hook up your server to the DNS name This has to be a valid domain and accessible from the outside world. You then open port 5060 for protocol UDP (very important, this is NOT the same as TCP!) on your outside firewall or server and connect that to the virtual PBX. This allows telephones to register themselves from remote locations such as home or a 2nd office but also allows you to call straight into the office.

The details of this I still have to work out (I’ll update this article as I go) but it comes down to this: the caller calls This email address is being protected from spambots. You need JavaScript enabled to view it.. His PBX will know that it has to connect to on port 5060 and ask the PBX at that address to ring up number 1234. The office PBX recognises that the incoming call request is not from a registered internal extension and determines how the call should be handled. If all goes well, the right trunk will be activated which in turn will ring the extensions you have linked to that line. This way, the POTS system is not used at all, and as another benefit, the call is absolutely free as only the internet is used.

I hope this helps others out there to determine which hardware they should buy and how most things fit together. Most devices marketed as “VoIP adapters” are not quite accurate labelled as they normally have a small SIP server (making it in fact a mini-PBX) and one or more FXS and FXO ports. The type of ports restrict you to their usage and it is very important to decide what you need. If you want to hook up a number of old telephones you need FXS ports, if you want to connect your physical telephone line to your virtual PBX, you need an FXO port.

In another article I will continue my findings as I just ordered some VoIP hardware to start experimenting with it.