Posted: Sat Dec 20, 2003 8:17 pm Post subject: [Asterisk-doc] docs chapter07.xml,1.1,1.2
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 -> 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&type=bug">http://bugs.digium.com/file_download.php?file_id=448\&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>
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