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

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=11849 
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Reported By:                greyvoip
Assigned To:                mnicholson
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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:              2009-06-03 18:36 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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Relationships       ID      Summary
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related to          0013892 After upgrading from 1.4.21.2 to 1.4.22...
related to          0014398 Calls coming in then out do not get rec...
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 (0105960) sverre (reporter) - 2009-06-03 18:36
 https://issues.asterisk.org/view.php?id=11849#c105960 
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I mentioned it earlier but it's worth repeating. I have accounts with a
number of SIP providers that use Asterisk. I have verified that I can
currently make near free mobile or international calls by placing the call,
then doing a transfer to a local number. The timer for the mobile call
stops, and the local call is a flat rate.

I've heard stories of hacked Asterisk machines being used to place huge
numbers of expensive international calls, so there's clearly a motivation
for people with malicious intent to exploit this security hole. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-06-03 18:36 sverre         Note Added: 0105960                          
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