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

Danny Nicholas danny at debsinc.com
Thu Jan 24 13:21:00 CST 2013


This might have changed but IIRC /etc/asterisk/manager.conf controls what
events you have access to.

 

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at 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

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