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How to add Skype and FreeSwitch to existing PIAF 2013-05-25

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Update!

Skype static version 2 was removed from Arch Linux site so, I have create a new setup-skype-4.pl. This time will download v4.2.0.11 off Skype website. The rest of the step still still the same except you need to enable the sound back on. Just run setup-skype-4.pl instead of setup-skype-new.pl.

Goal:
- To integrate Skype to PIAF
- Ability to have multiple Skype channels with same username and multiple usernames.
- No need for dedicate server to run FreeSwitch

Credits/References:
1. Big thanks to Bill Simon who wrote This Guide
2. FreeSwitch WiKi
3. PIAF Members: Ward, w1ve, Geo...

Before You Start:
1. Make sure you have no sound driver loaded especially OSS by disable sound device in your bios or VM setting.
2. Run lsmod | grep snd ; You should see nothing
3. If you are on RentPBX, run yum install libXfont

Let's Get Started:
For those who use Ward VM with Skype support please skip install-freeswitch step, just replace freeswitch.xml and run setup-skype-new.pl

1. Download attach file, unrar and copy files to /root

2. Run
chmod +x /root/install-freeswitch /root/setup-skype-new.pl
/root/install-freeswitch
3. Let the script running, come back in about 15 mins and follow on-screen Skype channel setup

4. Run
cp /root/freeswitch.xml /usr/local/freeswitch/conf
pkill skype
pkill Xvfb
sh /usr/local/freeswitch/skypopen/skype-clients-startup-dir/start_skype_clients.sh
5. Wait for a few minutes and then login to your Skype desktop or mobile with another username and make sure Skype username that you set up for Freeswitch is not offline (Green or Yellow is fine)

6. Set up Trunk in FreePBX (Connectivity->Trunks)
Trunk name: freeswitch
username=freeswitch
type=peer
trustrpid=yes
sendrpid=yes
port=5070
insecure=port,invite
host=127.0.0.1
context=from-trunk
7. Setup Inbound route in FreePBX (Connectivity->Inbound Routes)
Description: Whatever name you want
DID number: Your Skype Username or Whatever you put in the setup process
Set Destination: Pick one from drop down list
8. Set up Outbound route in FreePBX (Connectivity->Outbound Routes)
Route name: Skype
Dial Pattern: I use *.
Trunk Sequence: Pick FreeSwitch Trunk from the list
9. Add local network to Asterisk Sip Setting (Settings->Asterisk Sip Settings)
Local Networks: 127.0.0.0/255.255.255.0
10. Run
/usr/local/freeswitch/bin/freeswitch
11. Make a test call

12. If you want to change user or number of channels, you can run /root/setup-skype-new.pl anytime.

13. If everything works fine then
echo "sh /usr/local/freeswitch/skypopen/skype-clients-startup-dir/start_skype_clients.sh"
>> /etc/rc.d/rc.local
echo "ulimit -s 240" >> /etc/rc.d/rc.local
echo "/usr/local/freeswitch/bin/freeswitch &" >> /etc/rc.d/rc.local
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sukasem
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Latest updates

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  3. Update Skype to Version 4.2.11

    Update Skype to V4.2.11