I think the problem I'm having is because I have also defined the reverse route (calls from PSTN to Asterisk), informing the Asterisk IP address in the "session target". So the IP is added to the "trusted list" and no authentication is required.
Does any one know how to force the digest authentication (as Asterisk does for SIP trunks type peer)?
But the problem is that the Cisco never Challenges the Asterisk (After receive the SIP Invite, the Cisco sends the 100 trying, then the 183 session progress, and then the call is established).
Maybe I'm missunderstunding somethinb because the only way I have found to get the calls from Asterisk to PSTN to work (without authentication) was informing the session target with the Asterisk IP in the dial-peer corresponding to the inbound leg, as follows:
dial-peer voice 2 voip description calls from Asterisk (inbound leg) session protocol sipv2 session target ipv4:184.108.40.206 incoming called-number . voice-class codec 1 dtmf-relay rtp-nte no vad ! dial-peer voice 4 pots description calls from Asterisk (outbound leg) destination-pattern . no digit-strip port 0/0/0:15
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