[Asterisk-Users] SIP Extension behind NAT, Asterisk on a public domain

[Tom Rymes]

On Nov 25, 2005, at 7:00 PM, Manny A. Wise wrote:

> -----Original Message-----
> From: Tom Rymes [mailto:trymes at cascadelinksystems.com]
[snip]
>> On Nov 23, 2005, at 11:04 AM, Manny A. Wise wrote:
>> [snip]
>>> Well, as the user stated on the original message, the asterisk
>>> server is behind a NAT and the client is also behind a NAT..
>>>
>>> if you make it work just by opening ports, let me know..I have
>>> never been able to get it to work, that's why I don't use sip, just
>>> plain iax2 for everything. J
>>>
>>> Manny
>>
>> Manny,
>>
>> I have this working as I write this. (I just hung up the phone.) In
>> fact, I brought a Cisco 7940G to a completely unknown nat-ed network
>> the other day, plugged it in and started making calls right away.
>> Here's the setup I have for this specific configuration:
>>
>> 1.) Asterisk server behind NAT. (Setup as DMZ on Linksys WRT54G, but
>> it's still NAT. I just don't have to forward ports this way)
>> 2.) externip, localnet, nat settings configured in the sip.conf file
>> (sip_nat.conf for Asterisk at Home)
>> 3.) Cisco phone (or whatever SIP UA you choose) configured for NAT
>> (via the SIP<MAC>.cnf file for Cisco)
>> 4.) Lather, rinse, repeat if necessary
>>
>> Hopefully that will work for you. I'd rather use IAX and avoid these
>> problems altogether, but I have yet to find an IAX hardphone I am
>> willing to use. In fact, for softphone use, I do indeed use IAX via
>> LoudHush for the mac. (Great piece of software, BTW. No connection
>> here, just a happy user...)
>>
>> Tom
>
> Great!!, this did the trick, now we have audio...
> We are using a Sipura 2000 for testing....
> The Sipura now can call out and have audio...the only problem left  
> is that
> the sipura can't receive calls, when the extension is dialed, the  
> recording
> says, the person is on the phone.....any ideas???
>
> I changed the externip=, localnet= and nat=yes in sip.com and in the
> extension setup in amp nat=1...... missing anything????
>
> THANKS!!!!!!!!!!!!!!!!
>
> Manny

It sounds as if your extension isn't registered. Make sure that the  
extension is configured as dynamic in sip.conf (or AMP) and as  
nat=yes. Also, make sure that the Sipura is configured through its  
web interface to register and it has the right user and password  
entered. Once this is done, when you type 'sip show peers' from the  
CLI your Sipura's extension should be listed, and show a 'D' and an  
'N' for dynamic and nat.

Also, it sounds like you are using AMP and or A at H, so make sure that  
you put the nat, externip, and localnet parameters in the  
sip_nat.conf file, *NOT* the sip.conf file, as that is likely to get  
overwritten by AMP. From my installation (obviously, substitute your  
external IP for the xxx.xxx.xxx.xxx below...):

[root at mercury root]# cat /etc/asterisk/sip_nat.conf
nat=yes
externip=xxx.xxx.xxx.xxx
localnet=10.0.0.0/255.255.255.0

Other than that, I recommend further google and voip-info spelunking  
expeditions to track down your problem. I think that voxilla.com also  
has good resources on the Sipuras

Tom

--------------------
Tom Rymes
Cascade Link Systems
www.cascadelinksystems.com
(603) 375-1414

"Intelligent technology solutions for small businesses."