kenmirrors/asterisk
1
0
Fork 0
mirror of https://github.com/asterisk/asterisk.git synced 2025-10-12 15:45:18 +00:00
Code Issues Packages Projects Releases Wiki Activity

Merged revisions 226531 via svnmerge from

Browse Source 
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/trunk@226532 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Joshua Colp
2009-10-29 18:13:42 +00:00
parent 64e8fb465b
commit b9c370da86
2 changed files with 10 additions and 2 deletions
Download Patch File Download Diff File Expand all files Collapse all files

4
doc/tex/localchannel.tex
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