[asterisk-users] identifying channel for softhangup

Danny Nicholas danny at debsinc.com
Tue Dec 29 14:23:33 CST 2009


Most of the asterisk-dev members read this discussion (In My Experience).
${EXTEN} in the case you state would be SIP/vgw1-00000075.  
Perhaps this link would be helpful
http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jeremy Kister
Sent: Tuesday, December 29, 2009 2:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] identifying channel for softhangup

On 12/29/2009 1:01 AM, Jeremy Kister wrote:
> e.g., in the first call, below, the channel name is 
> "SIP/vgw1-00000075" -- the second call (on the same FXO port after a 
> soft hangup on the CLI) is "SIP/vgw1-00000077"
> 
> How can I extract this information in the dialplan so that I can use 
> the SoftHangup app in asterisk to disrupt an existing call ?

can anyone think of a different mailing list which might have members 
who know the answer i'm looking for?  asterisk-dev ?

-- 

Jeremy Kister
http://jeremy.kister.net./

_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users




More information about the asterisk-users mailing list