[asterisk-bugs] [Asterisk 0017459]: Peer does not hang up when caller hangup while app_dial is executing - Deadagi

Asterisk Bug Tracker noreply at bugs.digium.com
Thu Jun 3 18:10:28 CDT 2010


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17459 
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Reported By:                mn3250
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17459
Category:                   Applications/app_dial
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.32 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-06-03 17:04 CDT
Last Modified:              2010-06-03 18:10 CDT
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Summary:                    Peer does not hang up when caller hangup while
app_dial is executing - Deadagi
Description: 
Looks like * does not hang up the dialed outgoing call even when an agi
hang up is issued.
====================================================================== 

---------------------------------------------------------------------- 
 (0122931) pabelanger (manager) - 2010-06-03 18:10
 https://issues.asterisk.org/view.php?id=17459#c122931 
---------------------------------------------------------------------- 
Thank you for taking the time to report this bug and helping to make
Asterisk better. 

Unfortunately, we cannot work on this bug because your description did not
include enough information. 

You may find it helpful to read the Asterisk Issue Guidelines
http://www.asterisk.org/developers/bug-guidelines. 

We would be grateful if you would then provide a more complete description
of the problem.

At a minimum, we need:
1. the specific steps or actions you took that caused you to encounter the
problem,
2. the behavior you expected, and
3. the behavior you actually encountered (in as much detail as possible).

This likely includes output from the console with debug level logging, a
SIP trace (if this is SIP related), and configuration information such as
dialplan (e.g. extensions.conf) and channel configuration (e.g. sip.conf).

Thanks! 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-06-03 18:10 pabelanger     Note Added: 0122931                          
======================================================================




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