[asterisk-commits] file: branch 1.6.2 r186635 - /branches/1.6.2/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Mon Apr 6 11:20:05 CDT 2009


Author: file
Date: Mon Apr  6 11:19:56 2009
New Revision: 186635

URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=186635
Log:
Blocked revisions 186624 via svnmerge

........
  r186624 | file | 2009-04-06 13:15:30 -0300 (Mon, 06 Apr 2009) | 13 lines
  
  Add support for changing the outbound codec on a SIP call using
  a dialplan variable.
  
  This adds a dialplan variable (SIP_CODEC_OUTBOUND) which controls
  the codec offered for an outgoing SIP call. This is much like the
  SIP_CODEC dialplan variable and has the same restrictions. The codec
  set must be one that is configured for the call.
  
  (closes issue #13243)
  Reported by: samdell3
  Patches:
        13243.diff uploaded by file (license 11)
........

Modified:
    branches/1.6.2/   (props changed)

Propchange: branches/1.6.2/
------------------------------------------------------------------------------
--- trunk-blocked (original)
+++ trunk-blocked Mon Apr  6 11:19:56 2009
@@ -1,1 +1,1 @@
-/trunk:182362,182521,182762,182960,183124,183148,183196,183239,183553-183555,184986,185299,185532,185581,185704,185741,185777,186078,186382,186525,186563,186566,186620
+/trunk:182362,182521,182762,182960,183124,183148,183196,183239,183553-183555,184986,185299,185532,185581,185704,185741,185777,186078,186382,186525,186563,186566,186620,186624




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