Note jitterbuffer support for chan_local in CHANGES

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85098 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago · 5aaaaed28d
parent 46b9ca721b
commit 5aaaaed28d
1 changed files with 9 additions and 3 deletions
--- a/6
+++ b/6
@ -289,3 +289,9 @@ Miscellaneous
     to just UNKNOWN if the extension exists.
  * 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.
+  * 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.