[Asterisk-Users] Asterisk behind NAT << How to do it.

[listas iPfone]

Hi

The version 1.260 of chan_sip.c already have that patch?:

http://bugs.digium.com/file_download.php?file_id=430&type=bug

thanks!

Miklos


----- Original Message ----- 
From: "Leif Madsen" <leif at hacklocalhost.com>
To: <asterisk-users at lists.digium.com>
Sent: Friday, November 28, 2003 2:10 AM
Subject: [Asterisk-Users] Asterisk behind NAT << How to do it.


> Thanks to ww and his patch on bug #104, I have successfully implemented
> Asterisk behind NAT without using STUN or anything crazy.  It's quite
> straight forward.
> 
> Until this gets tested enough and put into CVS, you will have to patch
> your chan_sip.c file to do this.  I'm sure within the next few days this
> will get put merged into CVS if no one finds any problems.
> 
> I tried this on chan_sip.c version 1.249 (the version the patch was
> written for) and the latest as of today 1.258.  Both work great.
> 
> Open ports 5060 and your RTP range (found in /etc/asterisk/rtp.conf). 
> Default is 10000 -> 20000
> 
> Forward ports 5060 and your RTP range to your internal Asterisk box.
> 
> For your sip.conf, you need to add three lines:
> 
> ; sip.conf snippet
> [general]
> port=5060                       ; make sure you have this line :)
> inside_net=192.168.1.100        ; this is the internal ip address of
> the                                ;
> asterisk server
> inside_mask=255.255.255.0       ; internal ip mask.  /24 as this example
> outside_addr=216.239.33.100     ; this can also be a FQDN! ie.
>                                 ; my.domain.com
> ; ... plus whatever else you have in your sip.conf
> 
> Download the patch at:
> http://bugs.digium.com/file_download.php?file_id=430&type=bug
> 
> Either update your Asterisk or verify you have at least version 1.249 of
> chan_sip.c:
> 
> cd /usr/src/asterisk/channels/
> cvs status chan_sip.c
> 
> ===================================================================
> File: chan_sip.c        Status: Locally Modified
>  
>    Working revision:    1.258
>    Repository revision: 1.258  
> /usr/cvsroot/asterisk/channels/chan_sip.c,v
> 
> While in pwd /usr/src/asterisk/channels/
> patch -p0 < /path/to/patch
> 
> Nothing should fail.
> 
> cd /usr/src/asterisk/
> make
> cp /usr/src/asterisk/channels/chan_sip.so /usr/lib/asterisk/modules/
> 
> Restart your Asterisk and try it.  If you want to call a NAT'd Asterisk
> box, my Free World Dialup number is 18924.  Currently online.
> 
> -- 
> Leif Madsen <leif at hacklocalhost.com>
> http://www.hacklocalhost.com
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>