[Asterisk-Users] Asterisk behind NAT << How to do it.
Balaji NJL
bajjeen at yahoo.com
Sun Jan 11 03:29:55 MST 2004
Hi All,
i just applied this patch. i need to test whether its
working. Can someone
connect to my server and leave me a vm at extension
2000.
Server : ojoobala.com
Phone
Extension : 2005
pwd : mytest
auth: md5.
pl leave a vm on extension 2000.
thanks a lot,
-B
----- Original Message -----
From: "listas iPfone" <listas at ipfone.com.br>
To: <asterisk-users at lists.digium.com>
Sent: Tuesday, December 09, 2003 2:10 AM
Subject: Re: [Asterisk-Users] Asterisk behind NAT <<
How to do it.
> 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
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> >
http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
>
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