[asterisk-bugs] [Asterisk 0013554]: Mixmonitor doens't record call after attended transfer (atxfer)

[Asterisk Bug Tracker]

The following issue has been SUBMITTED. 
====================================================================== 
http://bugs.digium.com/view.php?id=13554 
====================================================================== 
Reported By:                fabianoheringer
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   13554
Category:                   Applications/app_mixmonitor
Reproducibility:            always
Severity:                   feature
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.21.2 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             2008-09-24 15:18 CDT
Last Modified:              2008-09-24 15:18 CDT
====================================================================== 
Summary:                    Mixmonitor doens't record call after attended
transfer (atxfer)
Description: 
I don't know if is correct place to post this, please sorry if dont...
I'm using MixMonitor to record my outgoing and incoming calls, it's works
great, but if this call is transfered for another call via atxfer, when the
transfer is completed, the mixmonitor stop to record, my question it's a
bug or it's normal mixmonitor do that? If yes for the last, it's possible
to correct this?

below the log of an outgoing call, after answered by secretary, it was
transfered for another channel, see the moment when the call was transfered
and mixmonitor stopped to record:

    -- Starting simple switch on 'Zap/10-1'
    -- Executing [88037876 at Diretoria:1] Dial("Zap/10-1",
"DGV/g1/88037876||gtTM(grava|Rogerio|88037876)") in new stack
    -- Called g1/88037876
    -- DGV/1 answered Zap/10-1
    -- Executing [s at macro-grava:1] Set("DGV/1",
"CALLFILENAME=24092008-1718-Rogerio-88037876") in new stack
    -- Executing [s at macro-grava:2] MixMonitor("DGV/1",
"outgoing/24092008-1718-Rogerio-88037876.gsm|b") in new stack
  == Begin MixMonitor Recording DGV/1
    -- Started music on hold, class 'default', on DGV/1
    -- <Zap/10-1> Playing 'pbx-transfer' (language 'pt_BR')
    -- Executing [6 at Diretoria:1] NoCDR("Local/6 at Diretoria-ac7b,2", "") in
new stack
    -- Executing [6 at Diretoria:2] Dial("Local/6 at Diretoria-ac7b,2",
"Zap/5r2&Zap/6r2||tTg") in new stack
    -- Called 6r2
    -- Zap/6-1 is ringing
    -- Local/6 at Diretoria-ac7b,1 is ringing
    -- Zap/6-1 is ringing
    -- Zap/6-1 is ringing
    -- Zap/6-1 answered Local/6 at Diretoria-ac7b,2
    -- Stopped music on hold on DGV/1
    -- <Local/6 at Diretoria-ac7b,1> Playing 'beeperr' (language 'pt_BR')
  == End MixMonitor Recording DGV/1
  == Spawn extension (Diretoria, 88037876, 1) exited non-zero on
'Zap/10-1'
    -- Hungup 'Zap/10-1'

====================================================================== 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-09-24 15:18 fabianoheringerAsterisk Version          => 1.4.21.2        
2008-09-24 15:18 fabianoheringerSVN Branch (only for SVN checkouts, not tarball
releases) => N/A             
======================================================================