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Merged revisions 226532 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r226532 | file | 2009-10-29 15:13:42 -0300 (Thu, 29 Oct 2009) | 13 lines Merged revisions 226531 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226531 | file | 2009-10-29 15:11:26 -0300 (Thu, 29 Oct 2009) | 6 lines Add an option to enabling passing music on hold start and stop requests through instead of acting on them in chan_local. (closes issue #14709) Reported by: dimas ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@226534 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -27,6 +27,10 @@ audio that it receives from the channel that called the local channel. This is
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especially in the case of putting chan\_local in between an incoming SIP call
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and Asterisk applications, so that the incoming audio will be de-jittered.
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Using the "m" option will cause chan_local to forward music on hold start and stop
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requests. Normally chan_local acts on them and it is started or stopped on the
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Local channel itself.
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\subsection{Purpose}
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The Local channel construct can be used to establish dialing into any part of
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