[asterisk-bugs] [Asterisk 0017243]: attended transfered calls appearance in the queue_log

Asterisk Bug Tracker noreply at bugs.digium.com
Mon Apr 26 07:55:37 CDT 2010


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17243 
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Reported By:                Rami
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17243
Category:                   Applications/app_queue
Reproducibility:            always
Severity:                   trivial
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.30 
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:             2010-04-26 05:44 CDT
Last Modified:              2010-04-26 07:55 CDT
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Summary:                    attended transfered calls appearance in the
queue_log
Description: 
Hello,
When Agent receives a call from a queue and want to transfer it using
attended transfer the line which is written to the queue_log is incorrect,

The number the agent transfered the call to is not shown at all, instead
you get the number of the queue from where he received the call,
This makes it impossible to tell to what number the call was transfered to
as there are no more logs with the same uniqueid to tell us what happened
with the call.
====================================================================== 

---------------------------------------------------------------------- 
 (0120885) pabelanger (manager) - 2010-04-26 07:55
 https://issues.asterisk.org/view.php?id=17243#c120885 
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We need more information (IE: logs files and dialplans).
---
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\'d 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               
====================================================================== 
2010-04-26 07:55 pabelanger     Note Added: 0120885                          
======================================================================




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