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| author | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-10-29 18:18:30 +0000 |
|---|---|---|
| committer | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-10-29 18:18:30 +0000 |
| commit | ff4114bd3de1def9667f160869c1da801fbddf79 (patch) | |
| tree | dd7ac20c29311778d0acc235360bf17b0a857df6 /doc | |
| parent | 6dc887d1abb1448d18251f5b9ec248bf1ea7aa90 (diff) | |
Merged revisions 226532 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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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
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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
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@226540 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'doc')
| -rw-r--r-- | doc/tex/localchannel.tex | 4 |
1 files changed, 4 insertions, 0 deletions
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 |