[asterisk-users] question on SIP trunk and AMI to place call

[Jerry Geis]

I just put a break at dial_exec_full (app/app_dial.c for Asterisk 11.0.2)
did my AMI call

Action: Originate
Async: yes
Channel: SIP/testsystem/XXXXXXX

(calls from my machine over SIP trunk to another 11.0.2 box that has
a PRI card to make a call out to my cell)

and did not get a break.

Why is a SIP call not logging the Dial event as a DAHDI call does???

jerry