This Post is related to the original SipToSis with Asterisk from /dev/bios Blog
Nevertheless i finally managed to Get a working Skype <-> Asterisk connection via SipToSis. Hurray!
I didn’t get Skypeiax to work..
Hmm ok dann hoffen wir mal dass wir das auch hinbekommen 
This is how i did it - with Asterisk 1.4.x branch on the same machine skype should also do – running Debian 5.0
Ok ich setze mal auf Suse (10.3 produktiv, 11.2 als Testmaschine)
First you really need to have a working asterisk distribution , second you need to install Xvfb virtual frame buffer and skype itself the installation and configuration will depend between the systems so i can only cover debian here.
… und ich bin mal für die Suse Distri zuständig
Xtightvnc helps you to install and configure Skype the first time.
… und zum starten von Skype ohne Konsolen login
ok mal schauen ob wir dass später anders machen
- You need to install xvfb, tightvncserver, dbus-x11, dbus, xfce4 ( for configuring skype)
- Next, download skype (from here) and install it
- und für Suse nennen sich die Pakete – ein wenig anders
xvfb (Additional Xservers)
tightvnc (A virtual X-Window System Server)
dbus-1-x11 (D-Bus Message Bus System)
dbus-1 (D-Bus Message Bus System)
xfce4-desktop (Desktop Manager for Xfce) (for configuring skype) - Next, download skype (im Moment die 2.1 Beta) (from here) and install it (RPM installieren)
- Then we need SipToSis and install it:
Download it from here unpack it and move it into /usr/local/siptosis
Use the following command to startup an X-11 VNC Server to start skype and configure it:
# vncserver
Connect to vncserver, usually to the display “:1″ so in Tightvnc you need to enter the address like this: “172.0.0.254:1″.
After this, open up an console and start Skype:
# /usr/bin/skype
and configure a new (or existing) user in this VNC session, then close everything and kill the VNC Server:
#killall -9 vncserver
Now we have skype ready, we need to configure SipToSis, this is what i changed in the default siptosis.cfg config:
#host_port=5070 #username=skypests #passwd=unimportantpassword #do_register=no host_port=5070 contact_url=sip:skypetestuser@172.0.0.254:5070 from_url="skypetestuser" <sip:siptosisuser@172.0.0.254:5060> username=siptosisuser realm=asterisk passwd=siptosisregpassword expires=3600 do_register=yes minregrenewtime=120 regfailretrytime=15
Now we need to configure Asterisk with our new SIP User so it knows where to go when SipToSis tries to connect.
Asterisk’s sip.conf:
[siptosisuser] username=siptosisuser type=friend context=from-sip secret=siptosisregpassword host=dynamic nat=no dtmfmode=auto canreinvite=no ;(possibly set to yes if you know what you are doing) qualify=yes incominglimit=1 outgoinglimit=1 call-limit=1 busylevel=1 insecure=invite,port
As alwas the context is important, this is how i configured the extensions.conf:
For incoming Skype Calls:
[from-sip] ;Dialin from Skype exten => skype_input,1,Dial(SIP/myPhone,30,j)
For Outgoing Skype Calls:
exten => 101,1,Dial(SIP/siptosisuser/mySkypeUser)
Save and restart Asterisk.
Now it it time to boot everything together, but first we need a startup script for our SipToSys System.
Startup Script use this for easy copy & paste:
--------------------------------------------------
#!/bin/sh
# loading sound driver
/sbin/modprobe snd_dummy
# setting DISPLAY
export DISPLAY=:1
# starting virtual framebuffer X-Server
nohup /usr/bin/Xvfb ${DISPLAY} -screen scrn 300x600x8 >/dev/null &
#Start Skype
nohup skype >/dev/null &
#Give it some Time to boot
sleep 5
#Start SipToSiS
cd /usr/local/siptosis
nohup ./SipToSis_linux siptosis.cfg >/dev/null &
--------------------------------------------------
After executing it look in the asterisk console (asterisk -r) and try to call the Skype user!
Hopefully this is a way for you to make it easier to understand the Connection between Asterisk and skype.
Have phun 
References:
SipToSis Website: http://www.mhspot.com/sts/siptosis_ata_howto.html
SipToSis Forum: http://www.mhspot.com/stsforum/index.php
older SipToSis Infos: http://www.vitki.net/v/soho/voip
Startup Script: here
SipIAX Informations: http://www.opennet.ru/base/net/skype2sip.txt.html (use google translator)
Download it from here unpack it and move it into /usr/local/siptosis
