Merged revisions 228499 via svnmerge from

https://origsvn.digium.com/svn/asterisk/trunk

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  r228499 | file | 2009-11-06 13:52:00 -0400 (Fri, 06 Nov 2009) | 2 lines

  Fix the localchannel.tex file.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@228502 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/doc/tex/localchannel.tex b/doc/tex/localchannel.tex
index c8f6efb6221bff03ed3601f06be0bca440a71f58..d8651294ba8fb4a012feed1b0840407095f002b0 100644 (file)
--- a/doc/tex/localchannel.tex
+++ b/doc/tex/localchannel.tex
@@ -27,8 +27,8 @@ audio that it receives from the channel that called the local channel.  This is
 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}