[asterisk-users] Asterisk and NAT

[Vincent]

On Tue, 11 Dec 2007 11:05:24 -0600, Carlos Chavez
<cursor at telecomabmex.com> wrote:
>	The only thing you need to do is set nat=yes when you configure the
>phones in Asterisk.  You may need to use a STUN server in case the
>phones do not properly see the outside address.  Once the phones
>register they should be able to dial each other.

Unless I'm mistaken, there are other things to do:
- add the following in sip.conf
    [general] 
    ...
    externip=1.2.3.4 
    nat=yes 
    localnet=192.168.0.0/24 
    qualify=yes 
    canreinvite=no 
    
    [200] 
    ...
    qualify=yes ; Qualify peer is no more than 2000 ms away 
    nat=yes
    host=dynamic ; This device registers with us 
    canreinvite=no ; Asterisk by default tries to redirect 
    
- if the NAT firewalls that protect the two phones aren't
SIP-friendly, you'll have to force the phones to use specific UDP
ports for SIP and RTP, and map those ports on the firewall so that
incoming packets are let through.

I was about to post a message about Asterisk and NAT, but I might as
well ask here: Is there really no way to get two SIP devices to talk
to each other in case one or both are behind a NAT firewall, ie. RTP
packets must always go through Asterisk (hence, the "canreinvite=no")?

Thx
Fred.