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

[David J Carter]

Hi

I have applied the patch, I can register a Grandstream 100 from another
internet connection but I get no audio and a timeout line drop after 5
seconds.
If I call my SipPhone number 17476691936 I hear my welcome message and again
the line times out and drops after 5 seconds.
I notice that the connection is trying to do a native bridge even though I
have reinvite=no & canreinvite=no in the sip.conf.
Any help would be welcome.

Regards

Dave

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Leif Madsen
Sent: 12 December 2003 08:24
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Asterisk behind NAT << How to do it.

On Tue, 2003-12-09 at 05:10, listas iPfone wrote:
> 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

That link didn't work for me, but the NAT patch has not been put into
CVS yet.  It needs to be TESTED more, so if you guys want this added,
then you need to go and apply the patch and comment on it in the bug
tracker.  If something doesn't work, SAY SO!

Thanks :)

--
Leif Madsen <leif at hacklocalhost.com>
http://www.hacklocalhost.com
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