This might have changed but IIRC /etc/asterisk/manager.conf controls what events you have access to.
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Thursday, January 24, 2013 1:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] question on SIP trunk and AMI to place call You probably want the Dial event. It is raised both at the beginning of the Dial, as well as when the Dial completes. https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_Dial Note that the Channel: field will contain the name initiating the Dial, the Destination: field will contain the channel name being dialled, and the Dialstring: field will contain the non-technology specific portion of the thing being dialled. I get that even on the system with the PRI card and using DAHDI however I am not getting that event on the system with the SIP trunk . Is there something to enable to get that??? Both systems are running Asterisk 11.0.2. Thanks, 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
