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: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@226534 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Joshua Colp
2009-10-29 18:15:41 +00:00
parent 8ed14498d8
commit bf7f97a3ff
2 changed files with 10 additions and 2 deletions

View File

@@ -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