[asterisk-commits] file: branch 1.6.0 r188948 - in /branches/1.6.0: ./ channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: file
Date: Fri Apr 17 09:46:49 2009
New Revision: 188948

URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=188948
Log:
Merged revisions 188947 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

................
  r188947 | file | 2009-04-17 11:44:56 -0300 (Fri, 17 Apr 2009) | 22 lines
  
  Merged revisions 188946 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r188946 | file | 2009-04-17 11:41:25 -0300 (Fri, 17 Apr 2009) | 15 lines
    
    Fix a bug where a value used to create the channel name was bogus.
    
    This commit fixes the scenario where an incoming call is authenticated
    using a peer entry. Previously the channel name was created using either
    the username setting from the sip.conf entry or the IP address that the
    call came from. Now the channel name will be created using the peer name
    itself. This commit will not change the way the channel name is generated
    for users or friends.
    
    (closes issue #14256)
    Reported by: Nick_Lewis
    Patches:
          chan_sip.c-chname.patch uploaded by Nick (license 657)
    Tested by: Nick_Lewis, file
  ........
................

Modified:
    branches/1.6.0/   (props changed)
    branches/1.6.0/channels/chan_sip.c

Propchange: branches/1.6.0/
------------------------------------------------------------------------------
Binary property 'trunk-merged' - no diff available.

Modified: branches/1.6.0/channels/chan_sip.c
URL: http://svn.digium.com/svn-view/asterisk/branches/1.6.0/channels/chan_sip.c?view=diff&rev=188948&r1=188947&r2=188948
==============================================================================
--- branches/1.6.0/channels/chan_sip.c (original)
+++ branches/1.6.0/channels/chan_sip.c Fri Apr 17 09:46:49 2009
@@ -17305,7 +17305,7 @@
 
 			make_our_tag(p->tag, sizeof(p->tag));
 			/* First invitation - create the channel */
-			c = sip_new(p, AST_STATE_DOWN, S_OR(p->username, NULL));
+			c = sip_new(p, AST_STATE_DOWN, S_OR(p->peername, NULL));
 			*recount = 1;
 
 			/* Save Record-Route for any later requests we make on this dialogue */