[asterisk-users] Re: "Username/auth name mismatch" + SIP phone can't connect?

[Fred]

Hello,

Anselm Martin Hoffmeister > Try adding username=200 which fixed things for 
me. Alternatively, Try using a username that does NOT begin with a digit - 
I saw a flaky softphone some time ago that would screw completely with a 
numeric username.

Dovid B >The error you are getting is that asterisk has recieved the wrong 
user name  and or pass and is there for rejecting your registration. Your 
sip.conf  seems to be fine (although you may want to add dtmf and codec 
settings. Test  the same settings that you have now with a softphone and 
see if you recieve  the same errors or not.

Since the SJPhone could register OK (although sip debug showed some 401 at 
some point: maybe SJPhone supports some features that Asterisk doesn't, or 
at least are not supported by default), I figured it had something to do 
with the GrandStream phone : I had forgotten to turn off its use of NAT 
(STUN) :-/ This combined with a basic dial plan solved the issue.

For those interested, here's my basis sip.conf:
-------------------------- sip.conf -----------------------
[general]
context=default                 ; Default context for incoming calls
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

[200]
;username=200
type=friend
secret=test
qualify=yes ; Qualify peer is no more than 2000 ms away
nat=no ; This phone is not natted
host=dynamic ; This device registers with us
canreinvite=no ; Asterisk by default tries to redirect
context=internal ; the internal context controls what we can do

[201]
;username=201
type=friend
secret=test
qualify=yes ; Qualify peer is no more than 2000 ms away
nat=no ; This phone is not natted
;host=192.168.0.234
host=dynamic ; This device registers with us
canreinvite=no ; Asterisk by default tries to redirect
context=internal ; the internal context controls what we can do
-------------------------- sip.conf -----------------------

... and the basic extensions.conf:

-------------------------- extensions.conf -----------------------
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no

[globals]

[internal]
;BAD exten => ${EXTEN},1,Dial(SIP/${EXTEN})
exten => 200,1,Dial(SIP/200)
exten => 201,1,Dial(SIP/201)
exten => 202,1,Dial(SIP/202)
-------------------------- extensions.conf -----------------------

I'm pretty sure there's a way to simplify the above, but as you can see, my 
first attempt made Asterisk barf ;-)

Thanks guys for your help
Fred.