[asterisk-commits] file: branch 1.6.0 r228500 - in /branches/1.6.0: ./ doc/tex/localchannel.tex
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Nov 6 11:53:22 CST 2009
Author: file
Date: Fri Nov 6 11:53:18 2009
New Revision: 228500
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=228500
Log:
Merged revisions 228499 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
........
r228499 | file | 2009-11-06 13:52:00 -0400 (Fri, 06 Nov 2009) | 2 lines
Fix the localchannel.tex file.
........
Modified:
branches/1.6.0/ (props changed)
branches/1.6.0/doc/tex/localchannel.tex
Propchange: branches/1.6.0/
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Binary property 'trunk-merged' - no diff available.
Modified: branches/1.6.0/doc/tex/localchannel.tex
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.0/doc/tex/localchannel.tex?view=diff&rev=228500&r1=228499&r2=228500
==============================================================================
--- branches/1.6.0/doc/tex/localchannel.tex (original)
+++ branches/1.6.0/doc/tex/localchannel.tex Fri Nov 6 11:53:18 2009
@@ -27,8 +27,8 @@
especially in the case of putting chan\_local in between an incoming SIP call
and Asterisk applications, so that the incoming audio will be de-jittered.
-Using the "m" option will cause chan_local to forward music on hold start and stop
-requests. Normally chan_local acts on them and it is started or stopped on the
+Using the "m" option will cause chan\_local to forward music on hold start and stop
+requests. Normally chan\_local acts on them and it is started or stopped on the
Local channel itself.
\subsection{Purpose}
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