From 90d63f67afd40c087f60b7ea800bc284c1f24094 Mon Sep 17 00:00:00 2001 From: file Date: Thu, 29 Oct 2009 18:14:36 +0000 Subject: 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: http://svn.digium.com/svn/asterisk/branches/1.6.0@226533 f38db490-d61c-443f-a65b-d21fe96a405b --- doc/tex/localchannel.tex | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'doc') diff --git a/doc/tex/localchannel.tex b/doc/tex/localchannel.tex index 528421e0b..c8f6efb62 100644 --- a/doc/tex/localchannel.tex +++ b/doc/tex/localchannel.tex @@ -27,6 +27,10 @@ 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 +Local channel itself. + \subsection{Purpose} The Local channel construct can be used to establish dialing into any part of -- cgit v1.2.3