[Asterisk-doc] docs chapter07.xml,1.1,1.2

[blitzrage]

Comments:
Update of /cvsroot/asterisk/docs
In directory sc8-pr-cvs1:/tmp/cvs-serv16046/docs

Modified Files:
	chapter07.xml 
Log Message:
blitzrage
- add documentation for SIP behind NAT using the patch from bug #104
	(since it hasn't been included in CVS yet...)
Index: chapter07.xml
===================================================================
RCS file: /cvsroot/asterisk/docs/chapter07.xml,v
retrieving revision 1.1
retrieving revision 1.2
diff -C2 -d -r1.1 -r1.2
*** chapter07.xml	12 Dec 2003 05:14:15 -0000	1.1
--- chapter07.xml	20 Dec 2003 20:17:44 -0000	1.2
***************
*** 46,50 ****
  	<sect1>
  		<title>SIP and NAT</title>
! 		<para/>
  	</sect1>
  	<sect1>
--- 46,115 ----
  	<sect1>
  		<title>SIP and NAT</title>
! 		<para>
! 			At this time, Asterisk will NOT work "out of the box" behind NAT.
! 			However you can patch chan_sip which will allow SIP to work behind
! 			NAT much like Cisco does.  For the full history of the patch, you
! 			can view it at 
! 			<ulink url="http://bugs.digium.com/bug_view_page.php?bug_id=0000104">http://bugs.digium.com/bug_view_page.php?bug_id=0000104</ulink>
! 		</para>
! 		<para>
! 			On your router/firewall, you will need to open port 5060 and forward
! 			it to your Asterisk box to allow SIP messages through.  You will then
! 			need to open and forward your RTP ports to allow audio through (the ports
! 			are configurable in rtp.conf.  See chapter 3 for more information).
! 			The default RTP ports are the range of 10000 -&gt; 20000.
! 		</para>
! 		<para>
! 			In your sip.conf, you need to add three lines to your [general]	context.
! 			<programlisting>
! 			[general]
! 			port=5060 		<lineannotation>; make sure you have this line</lineannotation>
! 			externip=my.domain.com	<lineannotation>; this can be either external ip address, or FQDN</lineannotation>
! 			localmask=255.255.255.0	<lineannotation>; subnet mask.  This example is a /24 (class C)</lineannotation>
! 			localnet=192.168.0.0	<lineannotation>; local network your Asterisk server is in.</lineannotation>
! 			</programlisting>
! 		</para>
! 		<para>
! 			Now that we have our ports forwarded and our sip.conf file configured, we need to patch
! 			Asterisk so that this will all work.  First thing is to download the patch from
! 			<ulink url="http://bugs.digium.com/file_download.php?file_id=448&amp;type=bug">http://bugs.digium.com/file_download.php?file_id=448\&amp;type=bug</ulink> .
! 			Either update your Asterisk from CVS or verify that you have at least version 1.249 of
! 			chan_sip.c:
! 		</para>
! 		<para>
! 			<literallayout>
! 			<command>cd /usr/src/asterisk/channels/</command>
! 			<command>cvs status chan_sip.c</command>
! 			</literallayout>
! 		</para>
! 		<para>
! 			<programlisting>
! 			===================================================================
! 			File: chan_sip.c        Status: Locally Modified
!  
! 			   Working revision:    1.258
! 			   Repository revision: 1.258  
! 			/usr/cvsroot/asterisk/channels/chan_sip.c,v
! 			</programlisting>
! 		</para>
! 		<para>
! 			<literallayout>
! 			While in the present working directory /usr/src/asterisk/channels/
! 			<command>patch -p0 < /path/to/patch/chan_sip.c.1.259.diff</command>
! 			</literallayout>
! 		</para>
! 		<para>
! 			Nothing should fail.
! 		</para>
! 		<para>
! 			<literallayout>
! 			<command>cd /usr/src/asterisk/</command>
! 			<command>make</command>
! 			<command>cp /usr/src/asterisk/channels/chan_sip.so /usr/lib/asterisk/modules/</command>
! 			</literallayout>
! 		</para>
! 		<para>
! 			Restart your Asterisk and try it.
! 		</para>
  	</sect1>
  	<sect1>