Fix the localchannel.tex file.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228499 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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
+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
+requests. Normally chan\_local acts on them and it is started or stopped on the
 Local channel itself.
 
 \subsection{Purpose}