[Asterisk-Users] 'sip show users' shows NAT RFC3581

2006-03-30 Thread Douglas Garstang
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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Re: [Asterisk-Users] 'sip show users' shows NAT RFC3581

2006-03-30 Thread Aaron Daniel
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
[EMAIL PROTECTED]
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