[asterisk-bugs] [Asterisk 0016925]: [patch] app_queue: Log failed attempts to call members

Asterisk Bug Tracker noreply at bugs.digium.com
Mon Jun 14 15:43:01 CDT 2010


The following issue has been UPDATED. 
====================================================================== 
https://issues.asterisk.org/view.php?id=16925 
====================================================================== 
Reported By:                haakon
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   16925
Category:                   Applications/app_queue
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     confirmed
Asterisk Version:           SVN 
JIRA:                       SWP-999 
Regression:                 No 
Reviewboard Link:           https://reviewboard.asterisk.org/r/704/ 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!): 249233 
Request Review:              
====================================================================== 
Date Submitted:             2010-02-27 06:31 CST
Last Modified:              2010-06-14 15:43 CDT
====================================================================== 
Summary:                    [patch] app_queue: Log failed attempts to call
members
Description: 
This patch enables logging of all call attempts from a queue. Not only the
ones that do not fail.

The patch also introduces a new parameter "congestion" to both
RINGNOANSWER in queue_log and AgentRingNoAnswer AMI event, which is set to
1 if the call failed to go through because of technical difficulties.

This makes it easier to make queue_log statistics with information about
problems with an agent. For example if an agent has a faulty line, or your
telco/dahdi connection is having problems.

I am however unsure if everyone want this marked as an congestion from the
"AST_CONTROL_CONGESTION" frame. Since in my experience, this can come if a
SIP UA doesn't want to let you ring more than x seconds, etc. Most real
congestion problems come before this frame is generated. (read: before a
new channel is up at all) So if this patch should be applied, maybe it
should be configurable, or let out.
====================================================================== 

---------------------------------------------------------------------- 
 (0123392) pabelanger (manager) - 2010-06-14 15:43
 https://issues.asterisk.org/view.php?id=16925#c123392 
---------------------------------------------------------------------- 
Added to reviewboard. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-06-14 15:43 pabelanger     Reviewboard Link          =>
https://reviewboard.asterisk.org/r/704/
2010-06-14 15:43 pabelanger     Note Added: 0123392                          
2010-06-14 15:43 pabelanger     Description Updated                          
======================================================================




More information about the asterisk-bugs mailing list