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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
Asterisk Reference Documentation
--------------------------------
1) To generate a PDF from this documentation, you will need the rubber tool,
and all of its dependencies. The web site for this tool is:
http://www.pps.jussieu.fr/~beffara/soft/rubber/
Then, once this tool is installed, running "make pdf" will generate
the PDF automatically using this tool. The result will be asterisk.pdf.
NOTE: After installing rubber, you will need to re-run the top level
configure script. It checks to see if rubber is installed, so that the
asterisk.pdf Makefile target can produce a useful error message when it is
not installed.
2) To generate HTML from this documentation, you will need the latex2html tool,
and all of its dependencies. The web site for this tool is:
http://www.latex2html.org/
Then, once this tool is installed, running "make html" will generate the
HTML documentation. The result will be an asterisk directory full of
HTML files.