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 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
