Here’s a gross hack for whom is desperate to get it working on Gentoo.

Step 1 : Get the package

You can download the package from here.
Pick the package matching your own architecture.

Step 2 : Convert the DEB package and extract it

Install the package deb2targz if you don’t already have it.

The following command will create a tar.gz file from the .deb package :

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deb2targz spotify-client-qt_0.6.2.291.gcccc1f5.116-1_amd64.deb

You can then extract that tar.gz file like below :

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tar zxvf spotify-client-qt_0.6.2.291.gcccc1f5.116-1_amd64.tar.gz

This will create you a usr directory, containing spotify files.

Toss them at the appropriate places on your system :

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cd usr
mv bin/spotify /usr/bin/
mv share/pixmaps/* /usr/share/pixmaps/
mv share/spotify/ /usr/share/

Step 3 : Fixing library issues

Now this is the ugly part. Spotify is linked against specific versions of some libraries (SSL 0.9.8 and Crypto 0.9.8). In order to make it work, you need to fake those versions by creating a symlink from the wanted version to the existing (you’ll need to figure this out) version in your system.

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cd /usr/lib
ln -s libssl.so.1.0.0 libssl.so.0.9.8
ln -s libcrypto.so.1.0.0 libcrypto.so.0.9.8

Step 4 : Instructing firefox to open the spotify protocol with the Spotify application

You need to register the spotify protocol so that Firefox will know how to deal with those “spotify:” urls.

1. Open the about:config page
2. Create a boolean value by the name of network.protocol-handler.expose.spotify
3. Set its value to false
4. Click on a spotify link (for example a playlist link or such). A popup will open and ask you to pick an application. Use /usr/bin/spotify.

That’s it. All set.
I’m not proud of it…. but I wanted to use Spotify badly

Getting Spotify to run on Gentoo/Linux: A Gross and Cruel Hack is a post from: Tech@Sakana - A sysadmin's blog

1. The BLACKLIST function

Asterisk implements a BLACKLIST function which can be used to check whether a callerID is blacklisted or not, here is the synopsis (which you can get by connecting to thee asterisk server by issuing the asterisk -v command and then core show function BLACKLIST) :

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spaghetti*CLI> core show function BLACKLIST
 
  -= Info about function 'BLACKLIST' =- 
 
[Synopsis]
Check if the callerid is on the blacklist. 
 
[Description]
Uses astdb to check if the Caller*ID is in family 'blacklist'. Returns '1'
or '0'.
 
[Syntax]
BLACKLIST()
 
[Arguments]
Not available
 
[See Also]
DB
spaghetti*CLI>

This is what we’re going to use for this example.

2. Managing the blacklist

You can add callerIDs to the blacklist with the following command :

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spaghetti*CLI> database put blacklist 03001125000 "Mother-in-law from hell"
Updated database successfully
spaghetti*CLI> database put blacklist 03001125150 "FAX machine harassing me"
Updated database successfully

If later on you wish to remove an item from the blacklist (maybe the fax is not that bad after all !), use the following :

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spaghetti*CLI> database del blacklist 03001125150 
Database entry removed.

And of course you can list the content of the current blacklist like that :

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spaghetti*CLI> database show blacklist
/blacklist/03001125000                             : Mother-in-law from hell
1 results found.
spaghetti*CLI>

Use those commands to tailor the blacklist according to your needs.

3. Altering the dialplan to filter out the blacklisted numbers

Next you’ve got to update your incoming context to deal with blacklisted callerIDs. My incoming context is named “incoming-calls” but you might have to adjust that :

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[incoming-calls] ; on declare le contexte de reception d'appels depuis freephonie
exten => s,1,GotoIf($["${CALLERID(NUM)}" = "anonymous"] ? 10) 
exten => s,2,GotoIf(${BLACKLIST()} ? 10)
exten => s,3,Dial(SIP/phone1&SIP/phone2,10)
exten => s,4,Congestion()
exten => s,5,Hangup()
exten => s,10,Answer
exten => s,11,Wait(1)
exten => s,12,MusicOnHold(default)
exten => s,13,Hangup

So here, the rules 1 and 2 are to send both anonymous callers (mostly telemarketers over here) and blacklisted callers to an infinite holding music (starting at line 10). The rules 3,4,5 are to deal with other calls and direct them to 2 SIP phones.

That’s it ! Feel free to ask if you need more details about this set up.

Asterisk 101 : How to get rid of your mother-in-law … is a post from: Tech@Sakana - A sysadmin's blog

So this post will document how to sign up for a GV account as well as how to use it with Asterisk afterwards, in the prospect of using it if you are not in the USA.

In order to be able to sign up for GV, you need to meet 2 prerequisites :

1. You need to have a US IP address
2. You need to have a US phone number, which will be used to validate your GV account

Step 1 is left as an exercise to the reader (“Good luck ! I’m behind 7 proxies !” ).

Step 2 is the one we’re going to describe here, as an example of what you can pull with simple Asterisk configurations.

1. Getting a US phone number

What you really need to find is a DID (Direct Inward Dialing aka DDI : Direct Dial-In) provider. It will provide you with a US phone number and will patch that call either to another phone number or to a SIP address.

GV will only forward calls to US phone numbers, so you need one of those. IPKall is such a free provider. They will provide you an American phone number and will forward it to any SIP or IAX2 address. Creating an account only takes a minute.

Nothing fancy here : Simply fill the SIP account into the SIP phone number field, and the SIP servername or IP into the SIP proxy field.

2. Setting up Asterisk

Now IPKall will gladly forward the calls to your SIP account. You can setup this SIP account as simply as below :

IPKall has a specificity we need to take care of : calls can originate from 2 different servers. We need to have calls coming from both servers to get into the same context. To avoid repeating twice the same configuration block (with just the IP address being different), we’ll use a template.

We’ll also define a SIP account for a SIP phone (or for a softphone).

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[phone1]
type=friend
username=phone1
secret=SecretPassword
host=dynamic
context=myhouse
nat=yes ; if you need to connect from behind a nat
 
[ipkall](!) ; This section is only a template
type=peer
username=ipkall
context=ipkall-in
insecure=invite
 
[ipkall-01](ipkall) ; We import the previous template to setup calls from
host=66.54.140.46       ; the first IPKall server
 
[ipkall-02](ipkall) ; We import the previous template to setup calls from
host=66.54.140.47       ; the second IPKall server

I’ve setup an SIP “number” I’ve instructed IPKall to forward the calls to is “ipkall”. The configuration below must reflect this. Here, phone calls to the “ipkall” extension will be routed to the “phone1″ SIP phone.

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[ipkall-in]
exten => ipkall,1,Dial(SIP/phone1)
exten => ipkall,n,Hangup()

3.Setup a softphone and finally your GV account

Now all you need is a softphone (I like sflphone for that purpose) or a SIP phone to be configured as “phone1″ (check your favorite softphone instructions to do so) and head back to GV to create your account.

GV will assign you a phone number and will ask you to which phone number to route the phone calls. That’s where you’ll specify your IPKall phone number.

GV will then ask you to validate this phone call by giving you a 2 digit number and calling you on your IPKall phone number. You’ll have to pick up the call on your softphone and dial the 2 digit number.

Voila, the account is created and validated !

Any call to your GV phone number will be routed to IPKall which will send it to your Asterisk installation, which will then ring your softphone or SIP phone.

Pretty nifty, heh ?

Asterisk 101 – Ghetto GoogleVoice : Signing up for / using GV even if you’re not in the USA using Asterisk is a post from: Tech@Sakana - A sysadmin's blog

This snippet is to place into the extension.conf file :

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[incoming] ; incoming calls
exten => s,1,GotoIf($["${CALLERID(NUM)}" = "anonymous"] ? 10) 
exten => s,2,Dial(SIP/phone1&SIP/phone2,10)
exten => s,3,Congestion()
exten => s,4,Hangup()
exten => s,10,Answer
exten => s,11,Wait(1)
exten => s,12,MusicOnHold(default)
exten => s,13,Hangup

This block will check for the NUMber information of the Caller ID and if it is “anonymous”, it will send it to the step 10 which will pick up the call and start the holding music. If the Caller ID is not “anonymous”, it will ring 2 SIP phones for 10 secs, and if no one picks those, send it to congestion mode.

That’s all !

Edit : On a second thought, I think I’ll change that to have Asterisk to pick up the call and let the caller leave a message. That’s still going to catch those telemarketers (waste time & money), while giving a chance to legit callers to speak their message.

Asterisk 101 : How to troll telemarketers (aka automatically send hidden Caller ID to a waiting music forever) is a post from: Tech@Sakana - A sysadmin's blog

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;;;;;;;;;;;;;;;;;;;;;;;;;;
; Copy without formatting
;;;;;;;;;;;;;;;;;;;;;;;;;;
^Ins::
	Clipboard =    ; Purge Clipboard content
	Send ^{Ins}  ; Send the Ctrl+Insert command
	ClipWait        ; Wait till data is present in the Clipboard
	Clipboard = %clipboard%    ;remove formatting
Return

That’s it. When hitting CTRL-Insert, the selection will be copied into the clipboard without formatting.

I am aware there are other ways around this problem, most stripping the formatting when pasting, but I had little success with them.

What are your favorite AutoHotkey hacks ?

AutoHotkey : Copying without formatting is a post from: Tech@Sakana - A sysadmin's blog

This config sets up :

• SIP phones (for softphones or harware phones with SIP capabilities)
• Voice mails
• A few test phone numbers
• Forwarding of calls to a SIP provider for outbound and incoming calls (from/to PSTN)

That should be plenty already for a SOHO environment !

Note to French readers : Si votre FAI est Free, cette configuration fonctionne pour passer / recevoir des appels via le SIP de Free (Freephonie).

SIP configuration

SIP configuration (for softphones / SIP hardware phones, as well as outbound / incoming PSTN calls to/from an SIP-to-PSTN gateway provider) is performed in the /etc/asterisk/sip.conf file.

Here is an example :

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# cat sip.conf 
[general]
defaultexpiry=1800
dtmfmode=auto
qualify=yes
 
register => 09510XXXXX:ProviderPassword@freephonie.net
 
; selecting CODECs
disallow=all
allow=ulaw
allow=alaw
allow=speex
allow=gsm
 
language=fr
 
[freephonie] ; Outbound phone calls SIP configuration
type=friend
host=freephonie.net
username=09510XXXXX
fromuser=09510XXXXX
secret=ProviderPassword
nat=yes
insecure=port,invite
fromdomain=freephonie.net
context=fromfree
 
[client1] ; Softphone declaration
type=friend
username=client1
secret=Password01
host=dynamic
context=home
nat=yes ; connexion is allowed from NATted networks
 
[client2] ; 2nd softphone declaration
type=friend
username=client2
secret=Password02
host=dynamic
context=home
nat=yes

Dialplan configuration

Next you’ll need to setup a dialplan so that calls are routed as you wish.

The dialplan is setup in the /etc/asterisk/extensions.conf.

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; context for local SIP clients
[home] 
; if 101 is dialed then send it to SIP client1, wait 15 sec for a 
; pick up, then send it to voicemail
exten => 101,1,Dial(SIP/client1,15)
exten => 101,n,Wait(1)
exten => 101,n,VoiceMail(101@voicemails)
exten => 101,n,PlayBack(vm-goodbye)
exten => 101,n,Hangup()
 
; 201 phone number will lead to the 101 voicemail menu
exten => 201,1,VoiceMailMain(101@voicemails)
exten => 201,n,Hangup()
 
; same setup for 102 as for 101
exten => 102,1,Dial(SIP/client2,15)
exten => 102,n,Wait(1)
exten => 102,n,VoiceMail(102@voicemails)
exten => 102,n,PlayBack(vm-goodbye)
exten => 102,n,Hangup()
 
; 202 phone number will lead to the 102 voicemail menu
exten => 202,1,VoiceMailMain(102@voicemails)
exten => 202,n,Hangup()
 
; outbound calls :
; if dialing a 0 starting phone number, remove the leading 0 and patch it
; through the freephonie link (SIP-to-PSTN gateway service) 
exten => _0.,1,Dial(SIP/freephonie/${EXTEN:1}) 
 
; misc testing stuff
; when dialing 500, Asterisk will pick up and count 1,2,3 before hanging up
exten => 500,1,Answer
exten => 500,n,Wait(2)
exten => 500,n,SayDigits(123)
exten => 500,n,Hangup
 
; when dialing 501, Asterisk will pick up and repeat everything you say with
; as less delay as possible. Good to assess line latency.
exten => 501,1,Answer()
exten => 501,n,Playback(welcome)
exten => 501,n,Playback(demo-echotest)
exten => 501,n,Echo()
exten => 501,n,Playback(demo-echodone)
exten => 501,n,Playback(vm-goodbye)
exten => 501,n,Hangup()
 
[fromfree] ; Context of inbounds call coming from freephonie.net
; rings both client1 and client2 for 10 secs, then send it to congestion.
; congestion is when all the lines are busy.
; in this case freephonie.net will sent it to its voicemail system
exten => s,1,Dial(SIP/client1&SIP/client2,10)
exten => s,2,Congestion()
exten => s,3,Hangup()

Voicemail configuration

Finally, let’s see the voice mailboxes configuration. They are setup in the /etc/asterisk/voicemail.conf.

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[voicemails]
; 101 is the voicemailbox number, 1234 is the password, client1 is the matching client,
; and the email address is to whom to send the audiofile. 
101	=>	1234,client1,client1@sakana.fr
102	=>	5678,client2,client2@sakana.fr

Asterisk : Basic SOHO environment VoIP PABX configuration is a post from: Tech@Sakana - A sysadmin's blog

You can control this behaviour with this about:config property middlemouse.contentLoadURL. Set to true, the middle-click on the page will load the URL (if it is one) in the page, set to false it won’t.

Thanks Ayman !

Firefox (Linux) : “Middle-click on page loads URL” annoyance is a post from: Tech@Sakana - A sysadmin's blog

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/usr/lib/python2.6/site-packages/WebappConfig/content.py:27: DeprecationWarning: the md5 module is deprecated; use hashlib instead
  import md5, re, os, os.path
* Fatal error: Your configuration file sets the server type "Apache"
* Fatal error: but the corresponding package does not seem to be installed!
* Fatal error: Please "emerge >=www-servers/apache-1.3" or correct your settings.
* Fatal error(s) - aborting

You can solve that problem by editing the following file : /etc/vhosts/webapp-config and set vhost_server="lighttpd".

For further reference, you’ll also need the following USE flags for PHP :

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spaghetti etc # grep php /etc/portage/package.use 
dev-lang/php		cli cgi mysql xml session pcre sockets

The rest of the installation is pretty straight standard and is already well documented (here, there, or even there).

Gentoo : Running Cacti with LigHTTPD is a post from: Tech@Sakana - A sysadmin's blog

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$ xfconf-query -c xfce4-keyboard-shortcuts -np "/commands/custom/<SUPER>space" -t string -s "gmrun"

This example binds <WINDOWS>-<SPACE> to gmrun (which, by the way, is a great lightweight alternative to Launchy for Linux).

This bug is already filed in the bugtrack of Xfce, let’s hope it will soon be fixed .

Xfce4 : Bug with key bindings is a post from: Tech@Sakana - A sysadmin's blog

In this post I’ll describe my Conky setup and explain how to extend it to monitor your rTorrent downloads.

rTorrent is a great BitTorrent client which offers an XML-RPC interface to its core functions, making it easy to get the downloads status through scripting. You can read more about that in this previous post.

Starting by the end

Here is a screenshot of the final Conky layout we’re going to setup throughout this post.

There are actually 2 instances of Conky running simultaneously : one for the calendar widget on the left, and one for the system monitor containing the rTorrent status on the right.

The calendar widget

I can’t take credit for this calendar widget, which I’ve found in the forums. It took juste a little bit of tweaking to get to this Conky configuration file :

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own_window yes
own_window_type desktop
own_window_transparent yes
own_window_hints undecorated,below,skip_taskbar,skip_pager,sticky
 
update_interval 5
 
minimum_size 250
alignment ml
double_buffer yes
 
use_xft yes
xftfont DejaVu Sans Mono:size=20
#xftfont Bitstream Vera Sans:size=8
 
TEXT
$alignc${time %H:%M}
${color red}$hr$color
${time %A} ${time %d} ${time %B}
${color red}$hr$color
${execpi 300 cal | sed -e 's/'`date | awk '{print $3}'`'/\$\{color e84448}'`date | awk '{print $3}'`'\$\{color}/'}

execpi is the directive responsible for creating the calendar. It basically executes a shell one-liner which output is then used as a Conky configuration snippet. It runs at 300 seconds (5 minutes) intervals not to waste computing resources.

This is the end result :

The system monitor

The system monitor requires a basic Conky configuration file and a perl script.

Conky configuration file for the system monitor

The system monitor Conky configuration file is a simple one, with standard Conky elements. The only “original part” is the one collecting rTorrent information : like for the calendar widget, an execpi directive is used to run the “rtorrent-status.pl” every minute (60 seconds).

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own_window yes
own_window_type desktop
own_window_transparent yes
own_window_hints undecorated,below,skip_taskbar,skip_pager,sticky
 
update_interval 5
text_buffer_size 512
 
minimum_size 250
alignment mr
double_buffer yes
 
use_xft yes
#xftfont Fixed:size=12
xftfont Bitstream Vera Sans:size=8
 
TEXT
${color red}CPU$color       ${hwmon 0 fan 2} RPM
Core0  ${cpubar cpu1 4,200} @ ${freq 1} MHz ${hwmon 1 temp 1}C
Core1  ${cpubar cpu2 4,200} @ ${freq 2} MHz ${hwmon 2 temp 1}C
Core2  ${cpubar cpu3 4,200} @ ${freq 3} MHz ${hwmon 3 temp 1}C
Core3  ${cpubar cpu4 4,200} @ ${freq 4} MHz ${hwmon 4 temp 1}C
 
${color red}MEMORY$color
Memory ${goto 60} ${membar 4,200} ${memperc}%
Swap   ${goto 60} ${swapbar 4,200} ${swapperc}%
 
${color red}HDD$color
sda    ${diskio sda} ${alignc 50}${hddtemp /dev/sda}
 
${color red}NET$color
Down   ${downspeedf eth0} KiB/s
Up     ${upspeedf eth0} KiB/s
 
${color red}BitTorrent$color
${execpi 60 ~/bin/rtorrent-status.pl}

Perl Script to collect rTorrent download information

The following script pulls the information from the remote rTorrent client and outputs it as Conky configuration elements which are used by Conky through the execpi directive.

This script requires the Frontier::Client perl module, which I automatically got when I installed the xmlrpc-c package (I use gentoo and I had to set the “tools” USE flag for this package. If you use a different Linux distro, you might have to sort out this point).

It also require that rTorrent be installed/configured with XML-RPC enabled, and that the XML-RPC interface be exported through a HTTP server. It seems like a lot of work, but it is not if you follow those instructions.

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#! /usr/bin/perl
 
use strict;
use warnings;
 
use Frontier::Client;
use Data::Dumper;
 
# Configuration
 
use constant server		=> 'http://spaghetti/RPC2';
use constant bt_stats	=> '/tmp/rtorrent_data';
 
# Do not edit below this comment
 
my $server = Frontier::Client->new(url => server);
my $result = $server->call('d.multicall', "main", 
	"d.get_base_filename=",
	"d.get_bytes_done=",
	"d.get_size_bytes=",
	"to_kb=\$d.get_down_rate=");
 
open DATA, ">" . bt_stats;
 
my $idx = 1;
my @res;
foreach my $d (@$result) {
	my $cur = $d->[1];
	my $max = $d->[2];
	my $dl_rate = $d->[3];
	print DATA $cur/$max, "\n";
	push @res,
		sprintf("%.40s \${alignr}%7.1f KiB/s\$alignr\n" . 
			" " x 5 . "%3.2f%%" . "\${goto 70}\${execbar sed -ne '$idx p' " .
			bt_stats . '}', $d->[0], $dl_rate, 100 * $cur / $max);
	$idx++;
}
close DATA;
 
print join("\n\n", @res), "\n";

Here is the end result for this :

Bringing it all together

Finally I wrote this shell script to start both Conky instances at once. You can add this script in the “autorun” or equivalent settings for your desktop environment.

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#! /bin/sh
 
conky -dc ~/.conkyrc
conky -dc ~/.conkyrc_cal

That’s all folks !

That was it. If you didn’t know how flexible and easy to extend Conky truly is, I hope this post will have helped you to get started writing your own Conky widgets and monitoring/extension scripts.

If you have any question or suggestion, please share them in the comments.

conky : integrating rTorrent downloads monitoring is a post from: Tech@Sakana - A sysadmin's blog