FreePBX

If you want to use epel repositories to install Skype and other programs you have to exclude some packages in order to avoid conflicts.

# nano /etc/yum.repos.d/epel.repo

[epel]
...
exclude=libresample,fail2ban,libsrtp,libsrtp-devel,nodejs,nodejs-devel,npm

1. Add epel and nux.ro repositories, don't forget to exclude conflicting packages (see the section above)
2. Install skype, java, x11 and dbus

   # yum install skype java xorg-x11-server-Xvfb x11vnc dbus-x11

3. Prepare logfile for SipToSis

   # mkdir /var/log/siptosip.log
   # chgrp asterisk /var/log/siptosip.log
   # chmod 660 /var/log/siptosip.log

4. Download and configure SipToSis

   # su - asterisk
   $ mkdir siptosis ; cd siptosis
   $ wget "http://latest-siptosis-package.tar.bz2"  # Look on Google for a website to download it
   $ tar xvjf latest-siptosis-package.tar.bz2
   $ nano siptosis.cfg
   
        ...
        #Sample AUTO config with NO registration
        #  username and password not important in this mode
        #  Set to available port to transport SIP messages on siptosis computer
        host_port=5070
        username=AsteriskUserForSiptosipTrunk
        passwd=AsteriskPasswordForSiptosipTrunk
        realm=asterisk # change this according to your PBX
        do_register=no
        # --- end of NO registration example ---
        ...
        
   $ nano SkypeToSipAuth.prop
   
        ...
        #Examples:
        *,sip:AsteriskUserForSiptosipTrunk@192.168.1.1:5060 # DON'T FORGET TO CHANGE USER, IP AND PORT ACCORDING TO YOUR ASTERISK CONF. For instance if you are using pjsip for extensions on port 5060 and sip for trunks on port 5061 YOU MUST use 5061 in this row otherwise SipToSis will try to authenticate itself calling the pjsip instance that doesn't know it as a trunk.
        ...
        
        ...
        #Default: all incoming skype callers get the invalid destination message
        #*,play:clips/invalidDest.wav
        ...

5. Create a start script

   $ cd
   $ nano startskype
   
        #!/bin/sh
   
        SKYPE_BINARY="/usr/bin/skype"
        SIPTOSIS_BINARY="/home/asterisk/siptosis/SipToSis_linux"
        
        /usr/bin/Xvfb :101 -ac &
        sleep 15
        DISPLAY=:101 pulseaudio &
        DISPLAY=:101 ${SKYPE_BINARY} &
        x11vnc -display :101 &
        DISPLAY=:101 ${SIPTOSIS_BINARY} > /var/log/siptosip.log &
   
   $ chmod +x startskype

6. Create a stop script

   $ nano stopskype
   
        #!/bin/sh
        
        killall "pulseaudio"
        killall "skype"
        killall "SipToSis_linux"
        killall "x11vnc"
        killall "Xvfb"
        
   $ chmod +x stopskype

7. Start Skype

   $ ./startskype

8. Connect to the X server with your favorite VNC client, login with your account (Don't forget to tick “Remember me”!) and authorize SipToSis plugin
9. Go to FreePBX trunks configuration and create a new trunk with the following parameters:

   GENERAL
   Trunk name: Skype
   Outbound CallerID: [yourSkypeUsername]-Skype
   
   SIP SETTINGS
   outgoing
   Trunk name: out-Skype
   PEER Details:
        username=AsteriskUserForSiptosipTrunk
        type=friend
        secret=AsteriskPasswordForSiptosipTrunk
        context=from-trunk
        host=127.0.0.1
        port=5070
        nat=never
        dtmfmode=auto
        canreinvite=no ;(possibly set to yes if you know what you are doing)
        insecure=port,invite
        qualify=yes ; optional
        incominglimit=1
        outgoinglimit=1
        call-limit=1
        busylevel=1
        permit=127.0.0.1/255.255.255.255&[your_ips]/255.255.255.255
   incoming
   User Context: Skype
   USER Details:
        host=192.168.122.154
        port=5070
        insecure=port,invite
        secret=AsteriskPasswordForSiptosipTrunk
        type=friend
        username=AsteriskUserForSiptosipTrunk
        user=AsteriskUserForSiptosipTrunk
        fromuser=AsteriskUserForSiptosipTrunk
        nat=no
        qualify=yes
        disallow=all
        allow=ulaw&alaw&ilbc&speex

10. Configure an outbound route

    ROUTE SETTINGS
    Route name: Skype
    Trunk Sequence for Matched Routes: Skype
    
    DIAL PATTERNS
    prefix: 9999

11. Apply the new configuration and try a test call to 9999[skype-username] (e.g. 9999echo123)

1. Create a custom extension with the following parameter:

   GENERAL
   User Extension: [number_that_you_will_call]
   Display Name: [skype-username-of-your-friend]
   Link to a Default User: None
   
   ADVANCED
   Dial: SIP/Skype/[skype-username-of-your-friend]

2. Apply the new configuration and try a test call to [number_that_you_will_call]