[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 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             2008-01-26 11:11 CST
Last Modified:              2008-10-30 10:01 CDT
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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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 (0094414) mdu113 (reporter) - 2008-10-30 10:01
 http://bugs.digium.com/view.php?id=11849#c94414 
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greyvoip, as far as I know TRANSFER_CONTEXT is only applicable for pbx DTMF
transfers and doesn't work for SIP transfers (at least I couldn't make it
working with SIP transfers).
Unlike you, I haven't seen how it works in other softswitches, but I
imagine that having a cdr per channel would pose a problem of identifying
which cdr belongs to which call and that's probably not going to be simple
as well.
Having said that, I would vote for any solution that would allow me to do
proper call accounting even if it means to redo my billing system and if
everybody agrees that cdr per channel is a way to go, so be it. It just
doesn't look like it's going to happen any time soon. 
So my suggestion is this. If it's so hard to get cdr stuff in asterisk
right for historical/technical/political/whatever reason then we could use
the minimally intrusive approach I mentioned above. Seriously what would
you prefer, waiting who knows how long for the proper solution or start
billing now? 

Issue History 
Date Modified    Username       Field                    Change               
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2008-10-30 10:01 mdu113         Note Added: 0094414                          
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