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