There is also this basic CLI Guide, but this needs to be tested as I have personally not done it myself:
dial-peer voice 1 voip description Skype Incoming voice-class codec 1 session protocol sipv2 session target sip-server incoming called-number <<---- SIP user here dtmf-relay rtp-nte no vad ! dial-peer voice 2 voip description Skype Outgoing destination-pattern .T voice-class codec 1 session protocol sipv2 session target sip-server dtmf-relay rtp-nte no vad ! ... sip-ua credentials username password realm sip.skype.com authentication username password registrar dns:sip.skype.com expires 3600 sip-server dns:sip.skype.com
The part I am not sure about is if you need to create a DN for this, most other ITSP's you have to, so I'm guess if this is the case it would look something like this:
description SIP Trunk Registration
hold-alert 30 originator
Again I will point out that the above may not be complete or correct, this is how it once worked and I guess these days CCA takes care of this stuff, and i am not sure if the DN is actually required with Skype either, it is there as "Just Incase" only, you may want to test the config first using the other stuff.
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The first time I followed the sipskype.pdf that was floating around from late last year.. it had some differences.. regardless i made the mistake of putting in my phone number instead of the skype number after configuring the sip profile. Now I'm never prompted again, so I'm not sure where in the CLI/CCA I'm supposed to change that from the main number to the skype number 99105xxxx
I've got it working now thanks to Marcus's video. Theres some tweaking and debugging I need to do as I can't get international calls working and instead of rolling over to voicemail its just going fast busy. But incoming calls and outgoing local calls are working now.
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