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authorfile <file@f38db490-d61c-443f-a65b-d21fe96a405b>2009-11-06 17:53:58 +0000
committerfile <file@f38db490-d61c-443f-a65b-d21fe96a405b>2009-11-06 17:53:58 +0000
commitcd08e61cd04aea0dc260a31270c4a8d8b4638e38 (patch)
treee650951798986540e3b44ac5b82dfc92865398c8
parent1f40a5103be5b2947e716e8fd20152d1e9d7e294 (diff)
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. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@228502 f38db490-d61c-443f-a65b-d21fe96a405b
-rw-r--r--doc/tex/localchannel.tex4
1 files changed, 2 insertions, 2 deletions
diff --git a/doc/tex/localchannel.tex b/doc/tex/localchannel.tex
index c8f6efb62..d8651294b 100644
--- 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}