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authorrussell <russell@f38db490-d61c-443f-a65b-d21fe96a405b>2007-10-09 15:10:14 +0000
committerrussell <russell@f38db490-d61c-443f-a65b-d21fe96a405b>2007-10-09 15:10:14 +0000
commit4c29367323959fe219d1657610eec6a236d8f67b (patch)
tree398e430f3c7ece34fd6d165b71c9811cab496e60 /doc
parent72f4e1120235627c54e8720163e805e87cb1b132 (diff)
Add jitterbuffer support for chan_local. To enable it, you use the 'j' option
in the Dial command. The 'j' option _must_ be used in conjunction with the 'n' option. This feature will allow you to use the existing jitterbuffer implementation to put a jitterbuffer on incoming SIP calls connecting to Asterisk applications by putting a local channel in the middle. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85097 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'doc')
-rw-r--r--doc/tex/localchannel.tex2
1 files changed, 2 insertions, 0 deletions
diff --git a/doc/tex/localchannel.tex b/doc/tex/localchannel.tex
index 76cf44566..2a018e2ee 100644
--- a/doc/tex/localchannel.tex
+++ b/doc/tex/localchannel.tex
@@ -9,6 +9,8 @@ chan\_local is a pseudo-channel. Use of this channel simply loops calls back int
Adding "/n" at the end of the string will make the Local channel not do a native transfer (the "n" stands for "n"o release) upon the remote end answering the line. This is an esoteric, but important feature if you expect the Local channel to handle calls exactly like a normal channel. If you do not have the "no release" feature set, then as soon as the destination (inside of the Local channel) answers the line, the variables and dial plan will revert back to that of the original call, and the Local channel will become a zombie and be removed from the active channels list. This is desirable in some circumstances, but can result in unexpected dialplan behavior if you are doing fancy things with variables in your call handling.
+There is another option that can be used with local channels, which is the "j" option. The "j" option must be used with the "n" option to make sure that the local channel does not get optimized out of the call. This option will enable a jitterbuffer on the local channel. The jitterbuffer will be used to de-jitter 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.
+
\subsection{Purpose}
The Local channel construct can be used to establish dialing into any part of the dialplan.