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--- software:freepbx [2016/05/11 09:50] – [epel repository] Michele Porelli
+++ software:freepbx [2017/02/15 08:09] (current) – [epel repository] Michele Porelli
@@ Line 3: / Line 3: @@
 ==== epel repository ====
 If you want to use epel repositories to install Skype and other programs you have to exclude some packages in order to avoid conflicts.
-<code>[epel]
+<code># nano /etc/yum.repos.d/epel.repo
+[epel]
 ...
-exclude=libresample,fail2ban,libsrtp,libsrtp-devel
+exclude=libresample,fail2ban,libsrtp,libsrtp-devel,nodejs,nodejs-devel,npm
 </code>
+==== Skype in/out calls ====
+  - Add [[https://fedoraproject.org/wiki/EPEL|epel]] and [[https://li.nux.ro/repos.html|nux.ro]] repositories, don't forget to exclude conflicting packages (see the section above)
+  - Install skype, java, x11 and dbus<code># yum install skype java xorg-x11-server-Xvfb x11vnc dbus-x11</code>
+  - Prepare logfile for SipToSis<code># mkdir /var/log/siptosip.log
+# chgrp asterisk /var/log/siptosip.log
+# chmod 660 /var/log/siptosip.log</code>
+  - Download and configure SipToSis<code># 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
+     ...</code>
+  - Create a start script<code>$ 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</code>
+  - Create a stop script<code>$ nano stopskype
+     #!/bin/sh
+     killall "pulseaudio"
+     killall "skype"
+     killall "SipToSis_linux"
+     killall "x11vnc"
+     killall "Xvfb"
+$ chmod +x stopskype</code>
+  - Start Skype<code>$ ./startskype</code>
+  - 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
+  - Go to FreePBX trunks configuration and create a new trunk with the following parameters:<code>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</code>
+  - Configure an outbound route<code>ROUTE SETTINGS
+Route name: Skype
+Trunk Sequence for Matched Routes: Skype
+DIAL PATTERNS
+prefix: 9999</code>
+  - Apply the new configuration and try a test call to 9999[skype-username] (e.g. 9999echo123)
+=== Special numbers to call directly a Skype username (useful for a quick dial or when you can write just numbers on your phone keyboard) ===
+  - Create a custom extension with the following parameter:<code>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]</code>
+  - Apply the new configuration and try a test call to [number_that_you_will_call]