[svn-commits] kpfleming: branch 1.6.2 r200727 - in /branches/1.6.2: ./ CHANGES

Mon Jun 15 20:05:10 CDT 2009

Author: kpfleming
Date: Mon Jun 15 20:05:06 2009
New Revision: 200727

URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=200727
Log:
Merged revisions 200726 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

........
  r200726 | kpfleming | 2009-06-15 20:03:22 -0500 (Mon, 15 Jun 2009) | 6 lines
  
  Document the new automatic 'ignoresdpversion' behavior.
  
  Asterisk will now automatically ignore incorrect incoming SDP version numbers
  when necessary to complete a T.38 re-INVITE operation.
........

Modified:
    branches/1.6.2/   (props changed)
    branches/1.6.2/CHANGES

Propchange: branches/1.6.2/
------------------------------------------------------------------------------
Binary property 'trunk-merged' - no diff available.

Modified: branches/1.6.2/CHANGES
URL: http://svn.asterisk.org/svn-view/asterisk/branches/1.6.2/CHANGES?view=diff&rev=200727&r1=200726&r2=200727
==============================================================================

--- branches/1.6.2/CHANGES (original)
+++ branches/1.6.2/CHANGES Mon Jun 15 20:05:06 2009
@@ -40,7 +40,12 @@
    version received is different from the current SDP session version.  This
    option is required to interoperate with devices that have non-standard SDP
    session version implementations (observed with Microsoft OCS).  This option
-   is disabled by default.
+   is disabled by default. In addition, this behavior is automatic when the SDP received
+   is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
+   since the call will fail if Asterisk does not process the incoming SDP, Asterisk
+   will accept the SDP even if the SDP version number is not properly incremented,
+   but will generate a warning in the log indicating that the SIP peer that sent
+   the SDP should have the 'ignoresdpversion' option set.
  * The parsing of register => lines in sip.conf has been modified to allow a port
    to be present in the "user" portion. Please see the sip.conf.sample file for more
    information


