[asterisk-users] SIP registration

[Nathan Bell]

When my SIP phones try to register with my asterisk box, this is what I 
get my log file:

Mar 26 14:46:41 NOTICE[3896] chan_sip.c: Registration from 
'<sip:201 at 192.168.2.13>' failed for '192.168.3.2' - Not a local SIP domain

In sip.conf I have this for my global settings:
[general]
context=from-sip                ; Default context for incoming calls
                                ; if asterisk was compiled with OSP support.
realm=actarg.com                ; Realm for digest authentication
                                ; defaults to "asterisk"
                                ; Realms MUST be globally unique 
according to RFC 3261
                                ; Set this to your host name or domain name
bindport=5060                   ; UDP Port to bind to (SIP standard port 
is 5060)
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds 
to all)
srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
                                ; Note: Asterisk only uses the first host
                                ; in SRV records
                                ; Disabling DNS SRV lookups disables the
                                ; ability to place SIP calls based on domain
                                ; names to some other SIP users on the 
Internet
autodomain=yes                  ; Turn this on to have Asterisk add 
local host
                                ; name and local IP to domain list.
                                ; and multiline formatted headers for strict
localnet=192.168.2.0/23
qualify=no

And this for my local settings:
[201]
type=friend                    ; Friends place calls and receive calls
context=from-sip               ; Context for incoming calls from this user
secret=asteriskpassword
host=dynamic                   ; This peer register with us
callerid=John Doe <201>
disallow=all
allow=ulaw                     ; dtmfmode=inband only works with ulaw or 
alaw!
progressinband=no              ; Polycom phones don't work properly with 
"never"
dtmfmode=rfc2833               ; Choices are inband, rfc2833, or info
nat=no                         ; there is not NAT between phone and Asterisk
canreinvite=no                 ; disallow RTP voice traffic to bypass 
Asterisk

Is there a better way to do this? Am I missing something obvious?

Nathan Bell
IT Engineer
Action Target, Inc.