[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:                file
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Project:                    Asterisk
Issue ID:                   14628
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     closed
Asterisk Version:           SVN 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases):  1.4  
SVN Revision (number only!): 180682 
Request Review:              
Resolution:                 fixed
Fixed in Version:           
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Date Submitted:             2009-03-09 02:11 CDT
Last Modified:              2009-03-11 12:26 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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---------------------------------------------------------------------- 
 (0101563) svnbot (reporter) - 2009-03-11 12:26
 http://bugs.digium.com/view.php?id=14628#c101563 
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Repository: asterisk
Revision: 181345

_U  trunk/
U   trunk/channels/chan_sip.c

------------------------------------------------------------------------
r181345 | file | 2009-03-11 12:26:41 -0500 (Wed, 11 Mar 2009) | 21 lines

Merged revisions 181328 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r181328 | file | 2009-03-11 14:22:52 -0300 (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=181345 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-03-11 12:26 svnbot         Checkin                                      
2009-03-11 12:26 svnbot         Note Added: 0101563                          
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