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Note jitterbuffer support for chan_local in CHANGES

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85098 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.6.0
Russell Bryant 18 years ago
parent 46b9ca721b
commit 5aaaaed28d
1 changed files with 9 additions and 3 deletions
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CHANGES
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@ -289,3 +289,9 @@ Miscellaneous
to just UNKNOWN if the extension exists. to just UNKNOWN if the extension exists.
* When originating a call using AMI or pbx_spool that fails the reason for failure * When originating a call using AMI or pbx_spool that fails the reason for failure
will now be available in the failed extension using the REASON dialplan variable. will now be available in the failed extension using the REASON dialplan variable.
* Added jitterbuffer support for chan_local. This allows you to use the
generic jitterbuffer on incoming calls going to Asterisk applications.
For example, this would allow you to use a jitterbuffer for an incoming
SIP call to Voicemail by putting a Local channel in the middle. This
feature is enabled by using the 'j' option in the Dial string to the Local
channel in conjunction with the existing 'n' option for local channels.

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