[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:              2008-11-17 09:03 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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 (0094939) frawd (reporter) - 2008-11-17 09:03
 http://bugs.digium.com/view.php?id=11849#c94939 
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Hi greyvoip, this patch was just an idea for a new function on cdr.c, to be
called whenever there is a transfer.

I didn't even test applying and compiling with it, and being just a
function, it doesn't do or correct anything. I will try to improve
(including channel.h is probably not the way to go if not already there)
and test that with SIP transfers (now that I know more or less where those
things happen) as soon as I have a bit of time (maybe next weekend).

Having already lost a project thanks to that issue and being unable to lie
to my clients for stupid trust issues, I had time to wait for some expert's
feedback (to check if I'm wasting my time or not), but apparently our
master is or seems dead. New projects are coming and I'll soon have to make
that issue a priority if I have brain enough to understand as much as Murf
what's going on with those CDRs.
In last resort, I heard that Freeswitch had a better channel-event system
but it would be a pain to switch for many reasons (ex: I really like
Asterisk's logo, and love that bad feeling when reading chan_sip.c). 

Issue History 
Date Modified    Username       Field                    Change               
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2008-11-17 09:03 frawd          Note Added: 0094939                          
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