[asterisk-bugs] [Asterisk 0014092]: bad callerid on incoming SIP transferred calls

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=14092 
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Reported By:                zerros
Assigned To:                
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Project:                    Asterisk
Issue ID:                   14092
Category:                   Applications/app_dial
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.4.22 
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:             2008-12-16 09:27 CST
Last Modified:              2008-12-16 11:25 CST
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Summary:                    bad callerid on incoming SIP transferred calls
Description: 
Hello,  forgive me in advance for my bad english. I'm a french people.

I have a problem with the atxfer features on incoming calls. If a incoming
calls is answerd by the receptionnist and she has to transfer the call
using atxfer, the callerid displayed on the final ST2030 is the number that
the external caller have dialed.

The final person can not know that it is a direct or a transferred call.

If I use blindxfer, the callerid is OK : the number of the original caller
is displayed on the phone.

I think there is a bug with atxfer. the callerid have to be the local
callerid of the receptionnist.

I have tried it on the 1.4.22 asterisk version.
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 (0096457) davidw (reporter) - 2008-12-16 11:25
 http://bugs.digium.com/view.php?id=14092#c96457 
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I think what is being said is that the final callee (party C) sees the
caller-id of the secretary (party B), which, in simple cases, is what the
external caller (party A) dialled, but may not be.

I'm not sure of the details for res_features transfers, but:

- for attended transfers in general, the caller-id is fixed at the time of
the enquiry, and it is not possible to know that this is actually an
attended transfer, rather than an enquiry;

- for SIP transfers, the enquiry is indistinguishable from a new call on a
second line on the the transferror's phone, as far as the PABX is concerned
(although the phone may instruct party C to take over the call between A
and B, that switching is handled within Asterisk, not passed to party C's
phone - Asterisk is a back to back user agent, and the conversation between
the PABX and party C is a separate SIP connection) - this means that if a
res_features transfer gave party A's caller-id, it would be inconsistent
with a SIP transfer;

Also, the same sequences, but more likely to be treated as an enquiry, are
used in broker situations, and, in that case, it may well be undesirable
for the target of the enquiry to know who the broker's client is. 

Issue History 
Date Modified    Username       Field                    Change               
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2008-12-16 11:25 davidw         Note Added: 0096457                          
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