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: https://origsvn.digium.com/svn/asterisk/trunk@85097 65c4cc65-6c06-0410-ace0-fbb531ad65f3