Debian 6 Squeeze(32/64) - Asterisk 11, FreeSWITCH living under one root(f) to make SKYPE free again!

Images taken off various sources on the internet

[UPDATE 15-10-2013]

This guide was taken off various sources but it all started from the Nerd Vittles’s forum entry here and various other wonderful resources such as FS-Wiki. I do not claim any rights/credits to this. It’s all of the contributors around the world of Asterisk that helped me put together a simple guide for simple implementation. Thanks to all the great work from all these great people who made this guide possible, Ward Mundy, the work from PSU VoIP, thank you. This is just a shortcut small mod for Debian 6 (Squeeze) 32/64bit.

Ready to Skype?

 
(img src rapgenius.com)

 

Warnings/Notices and Requirements:

• I STRONLY RECOMMEND USING 32 BIT SYSTEMS TO AVOID THE SKYPE CLIENT COMPLICATIONS WITH 64 BIT…

• WARNING, THIS TUTORIAL IS FOR EDUCATION PURPOSES ONLY

• WARNING, TEST THIS BEFORE IMPLEMENTING ON LIVE/PRODUCTION SYSTEMS

• IT IS IMPORTANT TO KNOW THAT EACH CHANNEL OPENS A NEW INSTANCE OF SKYPE CLIENT ON A WHOLE NEW VIRTUAL ENV, SO, HAVE LOTS OF MEMORY IF YOU WANT TO RUN MULTIPLE INSTANCES/CHANNELS

• I STRONGLY RECOMMEND YOU TO USE A PHYSICAL SERVER/PC TO DO THIS, VMS ARE FINE FOR TESTS!

• A sound card
• Virtualised sound card (like in VMs), like below for VirtualBox (to detect audio device in VMs, you need their respective guest addons software)
• You must be able to see an audio device when do you
  • #lspci | grep Audio
• e.g. Virtualbox setup for virtual audio
   
  • NOTE: I’ve had better quality with ICHAC97 for virtual box

Optionally you could

• Run a local asterisk (all in one soup)
• swsterisk debian 64bit image

How it works

• A Skype mod in FreeSWITCH called mod_skypopen load and registers Skype client connections made by the Linux Skype client which can be loaded as many times as you want (each load=1 channel) meaning you can possible have multiple channels on Skype for free

How is this different from Skype Manager/Business

• This is not the SIP trunk service called SkypeConnect
• It connects far faster than SkypeConnect
• It supports multichannels for Free
• It support all skype clients, including normal skype users or business users (created in Skype Manager)
• This is free but should be used for testing (do not infringe any terms and conditions from Skype by using this commercially)
• Its FREE
• Its FREE
• Its FREE again

IMPORTANT

• You will need to expose your skype password in clear, so be aware of that
• At minimum you need FreeSWITCH, Asterisk is optional, but since are Asterisk and FreePBX fanboys we make the all-in-one soup

And here are the steps!

• You can use a bare installed Debian 64 or 32 (highly recommended to use 32 bit)
• Add contrib into your repo
• #nano /etc/apt/sources.list
• At the end of each line that says main, add the word contrib like below
  
• Run
• #apt-get update 
• #apt-get install libX11-dev subversion automake autoconf wget  libtiff4-dev libtool \
  libncurses5-dev xvfb libx11-dev libasound2-dev xfs xfonts-100dpi \
  xfonts-75dpi xfonts-scalable git-core dpkg-cross binutils-multiarch libasound2 \
  libstdc++6 libgcc1 libstdc++6 libncurses5-dev zlib1g alsa-base \
  linux-sound-base libfontenc1 libfs6 libice6 libpixman-1-0 libsm6 \
  libx11-6 libx11-data libx11-dev libxau-dev libxau6 libxaw7 libxcb1 \
  libxcb1-dev libxcursor1 libxdmcp-dev libxdmcp6 libxext6 libxfixes3 \
  libxfont1 libxi6 libxinerama1 libxkbfile1 libxmu6 libxmuu1 libxpm4 \
  libxrandr2 libxrender1 libxss1 libxt6 libxv1 x11-common pulseaudio-module-hal     \
  x11proto-core-dev x11proto-input-dev x11proto-kb-dev xauth libqt4-dbus libqt4-webkit \
  xfonts-100dpi xfonts-75dpi xfonts-encodings xfonts-scalable \
  xfonts-utils xfs xkb-data xml-core xserver-common xtrans-dev xvfb \
  libgl1-mesa-dri libcurl4-openssl-dev libjpeg62-dev pulseaudio  xfonts-100dpi xfonts-75dpi xfonts-scalable xfonts-cyrillic \
  linux-headers-`uname -r` cabextract x-ttcidfont-conf ttf-mscorefonts-installer
• Select freetype when prompted
  • #dpkg-reconfigure x-ttcidfont-conf
• #nano /etc/X11/XF86Config-4
• Add  the following two lines
  • FontPath "/var/lib/defoma/x-ttcidfont-conf.d/dirs/TrueType"
  • FontPath "/var/lib/defoma/x-ttcidfont-conf.d/dirs/CID"
• #nano /etc/X11/fs/config
• At the end of the catalogue= add the following lines separated by commas like below (don’t forget the first comma)
  • ,/var/lib/defoma/x-ttcidfont-conf.d/dirs/TrueType/,/var/lib/defoma/x-ttcidfont-conf.d/dirs/CID/
• Restart xfs
  • # /etc/init.d/xfs restart
• FOR 64bit SQUEEZE only, NOTE: Because you are forcing different arcs, you may get errors on your next run of #apt-get install….
  • #apt-get install ia32-libs ia32-libs-gtk
  • #wget http://ftp.us.debian.org/debian/pool/main/q/qt4-x11/libqt4-webkit_4.6.3-4+squeeze1_i386.deb
  • #wget http://ftp.us.debian.org/debian/pool/main/q/qt4-x11/libqt4-phonon_4.6.3-4+squeeze1_i386.deb
  • #wget http://ftp.us.debian.org/debian/pool/main/q/qt4-x11/libqt4-xmlpatterns_4.6.3-4+squeeze1_i386.deb
  • #dpkg -i --force-architecture libqt4-webkit_4.6.3-4+squeeze1_i386.deb
  • #dpkg -i --force-architecture libqt4-phonon_4.6.3-4+squeeze1_i386.deb
  • #dpkg -i --force-architecture libqt4-xmlpatterns_4.6.3-4+squeeze1_i386.deb
  • Make sure you don’t have unfound dependencies
    • #ldd /usr/bin/skype  | grep not
    • You should see nothing
• #reboot    (strongly suggested)
• Time to install FreeSWITCH
• #cd /usr/src
• #git clone git://git.freeswitch.org/freeswitch.git
• #cd freeswitch && ./bootstrap.sh && ./configure
• #sed -i 's/\#codecs\/mod_g729/codecs\/mod_g729/g' /usr/src/freeswitch/modules.conf
• #sed -i 's/\#codecs\/mod_silk/codecs\/mod_silk/g' /usr/src/freeswitch/modules.conf
• #sed -i 's/\#endpoints\/mod_skypopen/endpoints\/mod_skypopen/g' /usr/src/freeswitch/modules.conf
• #make && make install
• #mv /usr/local/freeswitch/conf/autoload_configs /usr/local/freeswitch/conf/autoload_configs_noload
• #cd /usr/src/freeswitch/src/mod/endpoints/mod_skypopen/oss
• #make clean && make &&  insmod ./skypopen.ko && mknod /dev/dsp c 14 3

• Create a backup of the freeswitch.xml files as we will be getting our own basic configuration that does not listen to 5060 (As it will conflict with Asterisk, we will use UDP 5070) and simply throws all calls coming into FreeSWITCH to Asterisk via SIP and all calls from Asterisk to throw to mod_skypopen and into Skype VoIP cloud. If you are throwing the calls out on another Asterisk box (external box) be sure to modify the value  127.0.0.1 inside the new freeswitch.xml <settings> tag to something like your local IP and/or the public (natted) IP if it is natted. Also, when the asterisk server is external, change 127.0.0.1 to your external asterisk server IP address under the tag <gateways>. If you are planning to install and run Asterisk 11 on the same box, then don’t change anything there!

• #mv /usr/local/freeswitch/conf/freeswitch.xml /usr/local/freeswitch/conf/freeswitch.xml.bak
• #cd /usr/src
• #mkdir fs-skypecfg
• #cd fs-skypecfg
• #wget www.astiostech.com/public/fsskype/freeswitch.tar.gz
• #tar -zxvf freeswitch.tar.gz
• #cp ./freeswitch.xml /usr/local/freeswitch/conf/
• #chmod +x setup-skype-4.pl

• Ok, now its time to setup one or more of your Skype Accounts, simply run and follow on screen instructions. PS: If you can’t get through this below, what are you doing here mate?. WARNING! You should not get any errors! If you do, let us know report here!

• #./setup-skype-4.pl

  • It will ask your skype username

  • It will ask your skype password (be sure to test logon using a normal Skype client to see can work or not!)

  • Destination –> Use as suggested

  • Channels –> How many channels you need concurrently

  • Answer few more questions and it will download Skype client 4.2

  • At the end you must see the word SUCCESS!!!

• Be sure freeswitch run as a normal user for startime

  • #adduser --disabled-password  --quiet --system --home /usr/local/freeswitch --gecos "FreeSWITCH Voice Platform" --ingroup daemon freeswitch
  • #chown -R freeswitch:daemon /usr/local/freeswitch/
  • #chmod -R o-rwx /usr/local/freeswitch/

• Add the init scripts for freeswitch and fsskype as fsskype single script

  • #cd /etc/init.d/
  • #wget www.astiostech.com/public/fsskype/fsskype
  • #chmod +x fsskype

  • NOTE: fsskype starts and stops freeswitch as well, so just start fsskype during startup. Also be sure that this starts right at the end

  • #update-rc.d fsskype defaults 99

• Now, do your regular Asterisk install as described in the most basic way here, or if you are running Asterisk and FreePBX then great, proceed to enable/configure Asterisk trunk settings and allowing communications.

• Add a SIP trunk inside FreePBX like this

  • TrunkName: FreeSWITCH-Asterisk
  • Trunk Name in peer details: freeswitch
    • username=freeswitch
    • type=user
    • trustrpid=yes
    • sendrpid=yes
    • port=5070 ;remember freeswitch sip runs on this port
    • insecure=port,invite
    • host=127.0.0.1 ; remember freeswitch only listens to localhost
    • context=from-trunk
• Setup an inbound route. Lets say your skype username is sanjayws, then create an inbound route where the DID will then be sanjayws, e.g. like below
  
  • Send the incoming call to any destination of your fancy, in my case, its a conference room
• Under Asterisk SIP settings, be sure to allow 127.0.0.1 as your local network, to avoid one way audio issues or natting problems, my other network, the normal ETH0 network is 192….while the 202 is my liveIP incase i need to connect from Outside of my LAN or via my public network with NAT
  

• And you’re done!, lets start fsskype

  • WARNING you should not get any errors before, while or after starting fsskype!, if you do, report here

  • DO NOTE, the more channels you have the longer the start will be, it can be very long, so chillout and don’t freak out!

  • #/etc/init.d/fsskype start

• Now, you should see your skype username (if you’ve already added that person into your skype account, pop open as online). If you do not see it online, its likely that you’ve not logged into that account and added manually or just ignore that and straightaway right click on the name and call (not via PSTN call or Phone number)

• Just wait a while if Skype doesn’t automatically bridges calls to FS and Asterisk, wait like a minute!

• Try also outbound [not covering here], you need a phone capable of dialing out with letters since Skype are all alpha {duh}

• Well anyway, here are some screen shot

• My skype, with user sanjayW online inside our Debian box

  
  

• FreeSWITCH received the call and passing it to Asterisk

  

• Asterisk passing it to Conference app

  

• Process info – Skype single channel, if you got more channels, more lines will appear

  

 

Thank you for reading. And as usual please do give us your feedback! Thank you.