[asterisk-users] Asterisk 12 Dialplan

Murthy Gandikota mgandikota at nts.net
Mon Oct 27 10:56:01 CDT 2014


________________________________

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Richard
Mudgett
Sent: Friday, October 24, 2014 12:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 12 Dialplan

 

 

 

On Fri, Oct 24, 2014 at 1:19 PM, Murthy Gandikota <mgandikota at nts.net>
wrote:


In
https://wiki.asterisk.org/wiki/display/AST/Introduction+to+ARI+and+Chann
els

it is stated:

channel-dump.js in action

Here's sample output from channel-dump.js. When it first connects there
are no channels in Asterisk - (sad) - but afterwards a PJSIP channel
from Alice enters into extension 1000. This prints out all the
information about her channels. After hearing silence for a while, she
hangs up - and our script notifies us that her channel has left the
application.

<end of quote>
Is there some way the call can be moved to the next priority or context
in the dial plan from the stasis app? It seems the caller is stuck in
stasis.

 

Once a channel hangs up it is controlled by hangup handlers and h
extens.

If however you want to kick an active channel out of your stasis
application
to run dialplan then you use the

POST /channels/{channelId}/continue

ARI command.

 

Richard

[1]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Channels+REST+API
#Asterisk12ChannelsRESTAPI-continueInDialplan

 

 

Thanks, Richard. How do I get manager events such as  VarSetEvent
(https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+ManagerEvent_Var
Set) using ARI?

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