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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b><span style='font-weight:
bold'>On Behalf Of </span></b>Richard Mudgett<br>
<b><span style='font-weight:bold'>Sent:</span></b> Friday, October 24, 2014
12:02 PM<br>
<b><span style='font-weight:bold'>To:</span></b> Asterisk Users Mailing List -
Non-Commercial Discussion<br>
<b><span style='font-weight:bold'>Subject:</span></b> Re: [asterisk-users]
Asterisk 12 Dialplan</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
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12.0pt'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>On Fri, Oct 24, 2014 at 1:19 PM, Murthy Gandikota <<a
href="mailto:mgandikota@nts.net" target="_blank">mgandikota@nts.net</a>>
wrote:<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><br>
In<br>
<a
href="https://wiki.asterisk.org/wiki/display/AST/Introduction+to+ARI+and+Chann"
target="_blank">https://wiki.asterisk.org/wiki/display/AST/Introduction+to+ARI+and+Chann</a><br>
els<br>
<br>
it is stated:<br>
<br>
channel-dump.js in action<br>
<br>
Here's sample output from channel-dump.js. When it first connects there<br>
are no channels in Asterisk - (sad) - but afterwards a PJSIP channel<br>
from <st1:City w:st="on"><st1:place w:st="on">Alice</st1:place></st1:City>
enters into extension 1000. This prints out all the<br>
information about her channels. After hearing silence for a while, she<br>
hangs up - and our script notifies us that her channel has left the<br>
application.<br>
<br>
<end of quote><br>
Is there some way the call can be moved to the next priority or context<br>
in the dial plan from the stasis app? It seems the caller is stuck in<br>
stasis.<o:p></o:p></span></font></p>
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12.0pt'><o:p> </o:p></span></font></p>
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<p class=MsoNormal style='margin-bottom:12.0pt'><font size=3
face="Times New Roman"><span style='font-size:12.0pt'>Once a channel hangs up
it is controlled by hangup handlers and h extens.<o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>If however you want to kick an active channel out of your stasis
application<br>
to run dialplan then you use the<o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>POST /channels/{channelId}/continue<o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>ARI command.<o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>Richard<br>
<br>
[1] <a
href="https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Channels+REST+API#Asterisk12ChannelsRESTAPI-continueInDialplan">https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Channels+REST+API#Asterisk12ChannelsRESTAPI-continueInDialplan</a><o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 color=navy face="Times New Roman"><span
style='font-size:12.0pt;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Thanks, Richard. How do I get manager
events such as VarSetEvent (https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+ManagerEvent_VarSet)
using ARI?<o:p></o:p></span></font></p>
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