[Asterisk-Users] Re: Asterisk behind NAT << How to do it.(Leif Madsen)
David J Carter
david.carter at codepipe.com
Mon Dec 8 12:46:50 MST 2003
I have chan_sip.c version 1.259 do I still need the patch.
I can now get calls from sipphone.com but they drop after 5 seconds.
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Leif Madsen
Sent: 01 December 2003 18:39
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Re: Asterisk behind NAT << How to do it.(Leif
On Mon, 2003-12-01 at 05:52, Darren McIntosh wrote:
> In my configuration I have internal SIP clients registering from
> 192.168.0.0/28 and my * address is at 192.168.0.100. Using the host
> of the * box as the inside_net variable the audio from 192.168.0.0/28 was
> sent to the outside_addr variable giving one-way speech. Setting
> internal_net to the subnet address of 192.168.0.0 and inside_mask to
> 255.255.255.0 the call behaved correctly.
Aha! I had not tried this configuration. Now I see how that makes more
I will make note of that :)
Leif Madsen <leif at hacklocalhost.com>
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