[Asterisk-Users] read .what else to do ?

[Stewart Nelson]

Please note that recent IOS has SIP NAT traversal turned on by default.
I believe that it only supports internal UA / external server.
Since you also want the opposite, you should probably turn it off:
no ip nat service sip tcp port 5060
no ip nat service sip udp port 5060
Some IOS versions will even crash on SIP behind NAT.  See
http://lists.digium.com/pipermail/asterisk-users/2004-January/033718.html

Sorry, I don't know how to forward a range of ports.  To forward
a single port, use something like:
ip nat inside source static udp 192.168.1.2 5060 x.x.x.x 5060 extendable
where x.x.x.x is your public IP.
You can edit rtp.conf to use e.g 10000-10007 (would allow 4 calls) and
then only 8 ip nat statements would be needed for RTP.

You don't say what's failing.  "make calls outside our LAN" sounds like
you are trying to call using a VoIP provider that Asterisk registers
with.  But "your remote SIP phones" is something different; which of
the above are failing?  Are the registrations successful?  Is it just
the RTP that's not working (in which case the called phone will still
ring)?  If not, what error or timeout is reported?

If * verbose and/or debug logs don't show precisely what is going wrong,
use Ethereal (on both sides of the router if necessary) to see what
is happening.

--Stewart

>  Hi all ,
>   I have tried configuring Asterisk at home to make calls  outside our Lan
> WITHOUT any success (Setting up your router/firewall so  your remote SIP
> phones can communicate with your Asterisk at Home Server  via SIP through a
> NAT )
> 
>   To be precise i did the following
> 
>   (1) I Forwarded UDP Port 5060-5082 to 192.168.1.2
> Forward UDP Port 10000 to 20000 to 192.168.1.2
> 
>   (2) I set externip = x.x.x.x (to our public WAN)
>   localnet =192.168.1.0 /255.255.255.0
> 
>   (3) I also set nat=yes
>   qualify=yes
> 
>   (4)Please,I  know alot of you out there have implemented AAH to work
> outside your  network ( Setting up your router/firewall so your remote SIP
> phones can  communicate with your Asterisk at Home Server via SIP through a
> NAT  ).Please advise me how to make it work !!!
> 
>   (5) I am using xten lite soft phone on my pc .
> 
>   (6)  I use cisco 1700 series router ,and i have natting configured on
> this  router .Maybe I am using a wrong command .Please,tell me the
> commands  to forward the ports Port 5060-5082,10000 to 20000 to
> 192.168.1.2 on a  cisco router .
> 
>   Please reply and advice !!!
>   Thanks