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

[Matthew Jordan]

On 01/24/2013 01:13 PM, Jerry Geis wrote:
>>
>>
>> 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.
> 

The Dial events are created by app_dial. So long as you are using
app_dial to create your outbound channel, you should have that event.
Channel technology shouldn't matter.

-- 
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org