diff options
| author | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-10-29 18:13:42 +0000 |
|---|---|---|
| committer | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-10-29 18:13:42 +0000 |
| commit | 29706c54dfe735787b4d39ad9450903e8158bb39 (patch) | |
| tree | 1b9b5a693354ef4d76604687989b3fce3783ec2f /doc | |
| parent | 2bf61510dbe02860695da2c87cf2a8ffad89991d (diff) | |
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
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@226532 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 |