For 20090330 and earlier versions of stsTrunkBuilder, click here
Here are some Centos instructions for those who need a little extra help.
Install the applications
Install Asterisk, PBX-in-a-Flash, trixbox, Elastix or other SIP PBX.
Some tutorials are here: PBX-in-a-Flash without tears Elastix without tears trixbox without tears
Install Java 1.5 JRE or higher.
Download Skype client, install and create skype accounts for as many Skype channels as you need.
SipToSis must be run on the same computer as the Skype client.
Download the latest SipToSis SIP Skype Bridge and extract into a new folder 'siptosis'.
Download the latest SipToSis Skype Trunk Builder and extract into a new folder 'stsTrunkConfig'.
Make sure to keep the folder/path structure contained in the archives.
For linux only, install Xvfb if you don't already have it installed. (Example: yum install xorg-x11-server-Xvfb or sudo apt-get install Xvfb) - Some sort of X server is required.
Configure a functional test channel first.
It will be easier to correct a problem with a single channel. On linux: If you plan on using snd-dummy, test it thoroughly on this test channel.
Basic SipToSis setup is similiar to a SIP device - see the SipToSis ATA Setup Guide for initial setup.
This setup is expecting that you register SipToSis with the PBX.
Look for #Sample Asterisk registration example section in siptosis.cfg.
Below is a simple test Asterisk peer/extension setup. You need to set the desired context and set the registration info and IP address to match your SipToSis config.
For plain Asterisk:
Add to sip.conf
[siptosisuser]
username=siptosisuser
type=friend
context=default
secret=siptosisregpassword
host=dynamic
nat=no
dtmfmode=auto
canreinvite=no ;(possibly set to yes if you know what you are doing)
qualify=yes
defaultip=siptosisip
incominglimit=1
outgoinglimit=1
call-limit=1
busylevel=1
Add to extensions.conf:
exten => _7.,1,Dial(SIP/siptosisuser/${EXTEN:1})
;you would then dial 7 and the number you want to call
For FreePBX and variants (PBXinaflash/trixbox/Elastix): Screen shots
Create a SIP trunk:
Maximum Channels: 1
Dial Rules: .
Trunk Name: siptosisuser
Peer Details:
username=siptosisuser
type=friend
secret=siptosisregpassword
context=from-trunk
host=dynamic
nat=no
dtmfmode=auto
canreinvite=no ;(possibly set to yes if you know what you are doing)
qualify=yes
incominglimit=1
outgoinglimit=1
call-limit=1
busylevel=1
Create an Outbound Route:
Dial Patterns: 7|.
Trunk Sequence:
0 SIP/siptosisuser - click Add
Submit, Apply configuration changes.
Dial 7 and anything else from an extension and it calls out via the trunk.
For incoming calls, you should create a catch all inbound route and/or ring group. See screen shots Screen shots for example.
Dialing Note: Normal Asterisk PBX's can strip the prefix in extensions.conf file like this:
exten => _7X.,1,Dial(SIP/siptosisuser/${EXTEN:1}) or exten => _901.,n,Dial(SIP/stsTrunk_01/${EXTEN:3})
If your PBX can't be configured this way, then you can work around the PBX limitation by adding entries to SkypeOutDialingRules.props like this:
If your PBX dialing prefix is 7 you could add the following line:
^7([1-9][0-9]{10})$:+$1
The $1 will only capture what's in the parenthesis. In the example above, the 7 will be left out when making the SkypeOut call with 11 digit dialing. If 713051234567 was sent to the gateway, SipToSis would transform it to +13051234567. For other prefixes, change the 7 to the new prefix. Tip: make sure no other rules make the same match. Read up on regular expressions if you need to get more complicated.
After you have tested your single channel test configuration completely you are ready to build the multi channel trunk.
Scroll down to the Linux section or for Windows click here.