[asterisk-bugs] [Asterisk 0014628]: "SIP/2.0 404 Not found" when attended transferring a private number

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=14628 
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Reported By:                sverre
Assigned To:                
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Project:                    Asterisk
Issue ID:                   14628
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     ready for testing
Asterisk Version:           SVN 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases):  1.4  
SVN Revision (number only!): 180682 
Request Review:              
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Date Submitted:             2009-03-09 02:11 CDT
Last Modified:              2009-03-11 12:22 CDT
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Summary:                    "SIP/2.0 404 Not found" when attended transferring a
private number
Description: 
When a private incoming call arrives on Extension A as a private number,
the SIP Invite comes with "From: anonymous
<sip:None at 203.176.185.10>;tag=7c2dc3edeb530f73cd3bd6ae102a3d6b".

That call is placed on hold, and a new channel is opened between A->B.
When the attended transfer REFER command comes through, the following error
shows up in sip debug: "Failed SIP Transfer to non-existing extension None
in context phones", followed by Asterisk sending a SIP/2.0 404 Not found to
channel A.

The strange thing is that when the call ISN'T a private number, causing
the call to SIP invite to come with "From:
<sip:97891942 at 203.176.185.10>;tag=3c51826fb722e9d8fa4ee14f0e8d2bf4", the
SIP transfer is fine, "SIP transfer to extension 97891942 at phones by
100001102 at 192.168.0.21"

I tried adding an i extension to the phones context in the hope of
catching the None entry, but no luck.

While this was discovered in 1.4.22, it's present in SVN revision 180682,
and still applies after applying the patch found in bug
http://bugs.digium.com/view.php?id=14611

I've attached two console outputs with sip debug on, one version that
doesn't work with a private number, and one working with a normal CLID.
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---------------------------------------------------------------------- 
 (0101561) svnbot (reporter) - 2009-03-11 12:22
 http://bugs.digium.com/view.php?id=14628#c101561 
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Repository: asterisk
Revision: 181328

U   branches/1.4/channels/chan_sip.c

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r181328 | file | 2009-03-11 12:22:53 -0500 (Wed, 11 Mar 2009) | 14 lines

Fix issue where an attended transfer could not be completed under a rare
scenario.

When completing an attended transfer chan_sip does a check to make sure
the extension
in the URI portion of the Refer-To header is a local valid extension. We
don't actually
need to check this since we know for sure the other channel is already up
and talking to
the extension. Some devices do not put the extension in the Refer-To
header either, which
can cause the extension check to fail. We now no longer do this check if
it is an attended
transfer.

(closes issue http://bugs.digium.com/view.php?id=14628)
Reported by: sverre
Patches:
      14628.diff uploaded by file (license 11)

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http://svn.digium.com/view/asterisk?view=rev&revision=181328 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-03-11 12:22 svnbot         Checkin                                      
2009-03-11 12:22 svnbot         Note Added: 0101561                          
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