alang

You don't need to make any changes to the SipToSkypeAuth.props, just to keep it on the default settings.

Please be sure the line below existed in the SkypeOutDialingRules.props and the destination number is 55.

^55$:echo123

alang

I use asterisk -vvvvc because asterisk is not started after I logon to the computer.
I just don't kn ...
kurtor 發表於 2013-4-29 01:18

kurtor,
If you still didn't get it to work I really suggest you doing the setup from basis configuration, that means don't use the main configuration file 'siptosis.cfg' generated by stsTrunkBuilder unless you've realized every variables within the file.

Here are basis configuration for siptosis.cfg

host_port=5070
username=skypests
passwd=unimportantpassword
do_register=no

The proper trunk setting on asterisk/FreePBX

type=peer
host=<SipToSis LAN IP>
port=5070
nat=no
dtmfmode=auto
canreinvite=no
qualify=yes
incominglimit=1
outgoinglimit=1
call-limit=1
busylevel=1
context=from-trunk

Note: Be sure you've closed the SELinux & firewall on CentOS.

Good luck