[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/
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--- 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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