[asterisk-bugs] [Asterisk 0011849]: Missing CDR's for Transfers

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=11849 
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Reported By:                greyvoip
Assigned To:                murf
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Project:                    Asterisk
Issue ID:                   11849
Category:                   CDR/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     assigned
Asterisk Version:           1.4.17 
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:             2008-01-26 11:11 CST
Last Modified:              2009-02-25 07:58 CST
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Summary:                    Missing CDR's for Transfers
Description: 
At the moment there is one CDR generated per generic bridge. This tends not
to create any problems when the bridge has been created by something like:

SIP User -> Asterisk -> PSTN

The CDR generated will have the PSTN number as the destination and the SIP
User's accountcode.

When a transfer is undertaken the one CDR per generic bridge approach
breaks down. An example call flow for a blind transfer is:

SIP User -> Asterisk -> PSTN
PSTN <- Asterisk -> PSTN (this is after the user has blind transferred the
first call to a second PSTN number)

At the moment Asterisk will correctly generate a CDR for the first call
leg but for the second call leg there is a problem. For the sconed call leg
both ends of the bridge are now billable but as Asterisk only generates a
single CDR per bridge one of the legs will not get billed. 

A straight forward fix (at least architecturally) would be to generate a
CDR for each end of the bridge instead of combining both ends into a single
CDR. It would mean some extra CDR's for the standard SIP User -> PSTN call
but it's a lot easier to filter out CDR's to ignore than it is to try and
work out how to handle ones that are missing.

I've classified this as major since it's costing me (and other providers)
money every time a user does a transfer :).
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Relationships       ID      Summary
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related to          0013892 After upgrading from 1.4.21.2 to 1.4.22...
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 (0100704) sverre (reporter) - 2009-02-25 07:58
 http://bugs.digium.com/view.php?id=11849#c100704 
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I  have been carefully watching this thread recently, having come up
against the exact same issue. The thing is, blind transfers are not the
issue, putting one of the unique id's into the userfield allows the linking
of two CDRs that have been blind transferred.

The issue is only with attended transfers, because the moment BEFORE the
attended transfer takes place, they are two completely separate calls - two
unrelated CDRs in the making. The moment the SIP client sends the SIP REFER
however, one of the channels is dropped (ending the CDR timing on one of
the two), but no detectable event takes place on the Asterisk side that
allows you to set the userfield to the other channels uniqueid.

If there was just a variable ${ATTENDED_TRANSFER} made available to the h
extension in the same vein as ${BLIND_TRANSFER} is made available, then we
could solve the CDR problem at a dial plan level as a workaround until
NewCDR gets merged. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-02-25 07:58 sverre         Note Added: 0100704                          
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