[Asterisk-Users] Asterisk behind LinkSys NAT Routing

[Martin Pycko]

You can port forward the 5060 SIP port and use externip keyword in
sip.conf to have it working behind a NAT.

Martin

On Mon, 3 Nov 2003, WipeOut wrote:

> Robert Mann wrote:
>
> > Problem I have is this.  outside firewall (extension 2003) can call me
> > inside firewall (extension 2000) and all is fine.  If I call from
> > inside firewall (extension 2000) to outside firewall (extension 2003)
> > I hear no ringing and person at other end can pick up and I hear for
> > maybe a half second then I go to voicemail.  If I add another
> > extension on the outside then communication between outside and
> > outside through * is not possible at all.  I know I can not be the
> > only one who has tried to do this.  Please any help would be greatly
> > appreciated.
> >
>
> Robert,
>
> You need to get Asterisk onto a public IP address.. Using the DMZ
> function on the router will not work.. If you search the archives you
> will see that it has been attempted many times..
>
> The reason is not in the IP but in the SIP headers.. they will be sent
> out from the Asterisk server with the internal IP address of the server,
> this means that when the SIP UA reads the SIP message and responds it
> will respond to the incorrect IP address..
>
> So the basic rules where NAT is involved are..
>
> Asterisk server must always be on a public IP address..
>
> SIP UA's can be behind NAT but need "nat=yes", "canreinvite=no" and
> "qualify=yes" set in the phone configuration in sip.conf..
>
> Hope that helps..
>
> Later..
>
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