[asterisk-bugs] [Asterisk 0014660]: [patch] Lost info in CDR when transfering call via AMI's Redirect

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=14660 
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Reported By:                caspy
Assigned To:                mnicholson
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Project:                    Asterisk
Issue ID:                   14660
Category:                   CDR/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     ready for testing
Asterisk Version:           1.6.0.6 
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:             2009-03-13 09:11 CDT
Last Modified:              2009-08-20 12:59 CDT
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Summary:                    [patch] Lost info in CDR when transfering call via
AMI's Redirect
Description: 
- SIP/1344 call SIP/1340
- SIP/1340 answers, and they are speaking
- man-in-a-middle do a Redirect of SIP/1344 leg of this call to SIP/1341
- SIP/1341 answers, and 1344 speaks with 1341 successfuly.

The probles is in CDR.
For the first call (1344 to 1340) asterisk lost just all important
fields:


"","","s","fromoffice","","SIP/1340-08c6dce8","","","","2009-03-13
16:37:44","2009-03-13 16:37:47","2009-03-13
16:37:54",10,7,"ANSWERED","DOCUMENTATION","1236951464.550476",""

"","1344","1341","fromoffice","""Akzhan Test""
<1344>","SIP/1344-b4b3e758","SIP/1341-b51deec0","Dial","SIP/1341,60,twTWL(5400000)","2009-03-13
16:37:44","2009-03-13 16:38:00","2009-03-13
16:38:05",21,5,"ANSWERED","DOCUMENTATION","1236951464.550474",""


Also, mention that uniqueid is very strange: first part (time) is the
same, but second is decreesed for two. Why?
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 (0109354) escape2mtns (reporter) - 2009-08-20 12:59
 https://issues.asterisk.org/view.php?id=14660#c109354 
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Will do.  It will be later tonight but I can do it.

Our commands are going over the AJAM (not that it should matter).  The
outbound call goes out over a sip trunk to a landline (not that it should
matter either).

I'll let you know how it goes....  I may even try your test and see how it
works. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-08-20 12:59 escape2mtns    Note Added: 0109354                          
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