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Example Screenshots and sample config with FreePBX:
Trunk Setup Screen:
Here are the full peer details: username=skypetestuser type=friend context=from-trunk secret=skypetestuser host=dynamic nat=no dtmfmode=auto canreinvite=no insecure=invite qualify=yes ;defaultip= incominglimit=1 outgoinglimit=1 call-limit=1 busylevel=1 Outbound Route setup example:
My Asterisk Reg Section in siptosis.cfg for this example: #Sample Asterisk registration example host_port=5070 #contact_url is not needed unless ip address on siptosis console is incorrect - leave commented out. #contact_url=sip:skypetestuser@SipToSisIpAddress:SipToSisHostPort #from url is the sip address of the PBX from_url="skypetestuser" <sip:skypetestuser@192.168.0.7:5060> username=skypetestuser passwd=skypetestuser realm=asterisk expires=3600 do_register=yes minregrenewtime=120 regfailretrytime=15 For your system, set the IP address of the from_url above to be your PBX's IP. Change the realm if needed. With these settings siptosis should be able to register and you should be able to dial out using 7 as a dialing prefix. If the contact_url IP is incorrect on the siptosis console, contact_url in this example would be (assuming same box): contact_url=sip:skypetestuser@192.168.0.7:5070 Inbound Catch All Route Example - you decide what you want it to do (scroll down for a ring group setup example):
Ring Group Setup Example
You need to forward to the fake DID Number 123456789 that you setup on the Inbound Routes screen above. To do this, edit SkypeToSipAuth.props: *,sip:123456789@192.168.0.7:5060 Now all incoming Skype calls will go to the 601 ring group. |