[asterisk-bugs] [Asterisk 0015233]: MixMonitor stops after transfer from queue

Asterisk Bug Tracker noreply at bugs.digium.com
Tue Jun 16 18:37:32 CDT 2009


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=15233 
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Reported By:                jdennick
Assigned To:                lmadsen
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Project:                    Asterisk
Issue ID:                   15233
Category:                   Applications/app_mixmonitor
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.1.0 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-05-30 15:25 CDT
Last Modified:              2009-06-16 18:37 CDT
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Summary:                    MixMonitor stops after transfer from queue
Description: 
Exactly the same problem as was reported in Issue 0013538.  In direct
station-to-station or DAHDI-to-Station calls, recording will continue even
after a transfer.  In our situation, all inbound (DAHDI) calls are going to
a queue and then may need to be transferred to a Conference Room so others
can collaborate on the issue.  Obviously, we can record the Conference
(meetme) rooms individually, but it would be so much easier to just have
the original call recording contain the entire transcript.  I'll attach the
dial-plan in the "Additional Information" section.  The original recording
ends as soon as the Agent hits the pound (#) key to complete the blind
transfer (as configured in features.conf).
======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
related to          0014971 Monitor in Asterisk does not record all...
====================================================================== 

---------------------------------------------------------------------- 
 (0106526) jdennick (reporter) - 2009-06-16 18:37
 https://issues.asterisk.org/view.php?id=15233#c106526 
---------------------------------------------------------------------- 
The calls come into the queue directly from a T-1, so there isn't anything
happening before the dialplan that I presented above.  We are using Local
Channels for the queue agents (via RealTime), and I'll attach one of the
queue configs as well as one of the agent configs so you can see it:

Queue Config (from queue_table);
   name = 'KU'
   musicclass = 'beatles'
   announce = ''
   context = 'inbound'
   timeout = '12'
   monitor_type = ''
   monitor_form = ''
   queue_youarenext = ''
   queue_thereare = ''
   queue_callswaiting = ''
   queue_holdtime = ''
   queue_minutes = ''
   queue_seconds = ''
   queue_lessthan = ''
   queue_thankyou = ''
   queue_reporthold = ''
   announce_frequency = '0'
   announce_round_seconds = '0'
   announce_holdtime = 'no'
   retry = '5'
   wrapuptime = '15'
   maxlen = '0'
   servicelevel = '30'
   strategy = 'leastrecent'
   joinempty = ''
   leavewhenempty = ''
   eventmemberstatus = '0'
   eventwhencalled = '0'
   reportholdtime = '0'
   memberdelay = '0'
   weight = '0'
   timeoutrestart = '0'
   periodic_announce = ''
   periodic_announce_frequency = '0'
   ringinuse = 'no'
   setinterfacevar = '0'
   autopause = '1'
   
Agent (member) config (from queue_members table):
   +----------+------------+------------+-----------+---------+--------+
   | uniqueid | membername | queue_name | interface | penalty | paused |
   +----------+------------+------------+-----------+---------+--------+
   |      460 | Caeli      | KU         | SIP/2841  |       5 |      0 | 
   |      863 | Julie      | KU         | SIP/2842  |       4 |      0 | 
   |      295 | Ngoc       | KU         | SIP/2843  |       3 |      0 | 
   |      450 | Kathy      | KU         | SIP/2844  |       3 |      0 | 
   |      562 | Jenn       | KU         | SIP/2845  |       5 |      0 | 
   |      271 | Cris       | KU         | SIP/2846  |       1 |      0 | 
   |      301 | Sandy      | KU         | SIP/2847  |       1 |      0 | 

I think these are the answers to your questions, but if not, please do not
hesitate to ask for more details.  And thank you for your assistance in
this!

BTW: I was a good friend of Rich Adamson, in fact it was he who got me
involved with Asterisk in the first place.  This particular project is for
his company (Network Partners, Inc.).

Joe 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-06-16 18:37 jdennick       Note Added: 0106526                          
======================================================================




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