Files
asterisk/doc/tex
Joshua Colp 4eaa651a8a Add support for changing the outbound codec on a SIP call using
a dialplan variable.

This adds a dialplan variable (SIP_CODEC_OUTBOUND) which controls
the codec offered for an outgoing SIP call. This is much like the
SIP_CODEC dialplan variable and has the same restrictions. The codec
set must be one that is configured for the call.

(closes issue #13243)
Reported by: samdell3
Patches:
      13243.diff uploaded by file (license 11)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186624 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-06 16:15:30 +00:00
..
2008-07-31 16:37:08 +00:00
2009-01-21 22:04:16 +00:00
2008-07-31 16:37:08 +00:00

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.