asterisk users - Mar 2006 - 'sip show users' shows NAT RFC3581

Ok, this is highly confusing.

hestia*CLI> sip show users
Username                   Secret           Accountcode      Def.Context     
ACL  NAT
2944030                                     2944030          oneeighty_start  No
RFC3581
2944035                                     2944035          oneeighty_start  No
RFC3581

sip users (type=friend) are in sip.conf. I have nat=no against all of them. Why
does a 'sip show users' have RFC3581 against ALL my users? (there's
a lot more than I pasted here)

Thanks,
Doug.

Looked around a little.  If you set nat=never, then it won't set the 
phone to RFC3581... I haven't tested it, but you may want to try it :)

Aaron

http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html
"Nat
->This option determines the type of setting for users trying to connect 
to an asterisk server.

Possible values:


a) NAT=Yes, true, y, t, 1, on

All these values have the same behaviour, a combination of the options 
Route + rfc3581.

b) Nat=route:


Asterisk will send the audio to the port and ip where its receiving the 
audio from. Instead of relying on the addresses in the SIP and SDP 
messages.

This will only work if the phone behind nat send and receive audio on the 
same port and if they send and receive the signaling on the same port. 
(The signaling port does not have to be the same as the RTP audio port).

c) NAT=rfc3581

This is the default behaviour, is no nat=??? line is found for that user, 
this is the option used.

Asterisk will add an rport to the via header of the SIP messages, as 
described in rfc3581 (see http://www.faqs.org/rfcs/rfc3581.html), this 
will allow a client to request that the server send the response back to 
the source IP address and port where the request came from. The
"rport"
parameter is analogous to the "received" parameter in the VIA line,
except
"rport" contains a port number, not the IP address.

d) NAT=never

This will cause asterisk not to add an rport "rport" in the VIA line
of
the sip invite header, as introduced in rfc3581. (see 
http://www.faqs.org/rfcs/rfc3581.html) as some sip ua???s seem to have 
problems with them. (one of those UAs being the Uniden SIP phone UIP200 
??? Olle E. Johanson.)
"

On Thu, 30 Mar 2006, Douglas Garstang wrote:
> Ok, this is highly confusing.
>
> hestia*CLI> sip show users
> Username                   Secret           Accountcode      Def.Context   
ACL  NAT
> 2944030                                     2944030         
oneeighty_start  No   RFC3581
> 2944035                                     2944035         
oneeighty_start  No   RFC3581
>
> sip users (type=friend) are in sip.conf. I have nat=no against all of them.
Why does a 'sip show users' have RFC3581 against ALL my users?
(there's a lot more than I pasted here)
>
> Thanks,
> Doug.
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-- 
Aaron Daniel
Computer Systems Technician
Sam Houston State University
amdtech@shsu.edu
(936) 294-4198