asterisk users - Nov 2006 - "Username/auth name mismatch" + SIP phone can't connect?

Hello

	I'm trying to set up Asterisk on an older AMD Duron 700MHz with Fedora 5 
for use with SIP phones and the Linksys 3102 SIP gateway (ie. no FXO card, 
so no need for zaptel and libpri), but I'm stuck: The GrandStream BudgeTone 
phone fails registering with Asterisk :-/

Following the "Asterisk - The Future of Telephony.pdf", here's
what I did:

1. Installed Fedora 5, and ran "yum update", followed by "rpm
-Uvh kernel
kernel-devel" (yum would download the i686 version of "kernel"
but the i586
version of "kernel-devel"). I made sure it had all the requirements
for
Asterisk (ncurses + ncurses-devel, openssl + openssl-devel, zlib + 
zlib-devel, and bison)

2. Downloaded, unzipped, built, installed the following packages succesfully:
asterisk-1.2.13
asterisk-sounds-1.2.1

3. Edited /etc/asterisk/sip.conf thusly:
[200] ; extension 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

4. Added an empty section at the bottom of /etc/asterisk/extensions.conf:

[internal]
;later

5. Launched asterisk -vvvvvgc. Launches with no error message

6. Configured the GrandStream phone with user=200, authname=200, 
password=test, SIP server=192.168.0.252 (IP of Asterisk server), register=yes

7. When I plug/unplug the SIP phone, its network icon isn't displayed (ie. 
there's no connection with the SIP server, and no dial tone), and here's
what /var/log/asterisk/messages says:

Nov 13 19:52:33 NOTICE[17922] chan_sip.c: Registration from 
'<sip:200@192.168.0.252>' failed for '192.168.0.234' -
Username/auth name
mismatch

=> Is this due to wrong settings in sip.conf, the empty section in 
extensions.conf, the fact that I didn't add the Linksys gateway yet (how?), 
something else?

Thank you for any tip
Fred.

Am Montag, den 13.11.2006, 23:42 +0100 schrieb Fred:
> Hello
> 
> 	I'm trying to set up Asterisk on an older AMD Duron 700MHz with Fedora
5
> for use with SIP phones and the Linksys 3102 SIP gateway (ie. no FXO card, 
> so no need for zaptel and libpri), but I'm stuck: The GrandStream
BudgeTone
> phone fails registering with Asterisk :-/
> 
> Following the "Asterisk - The Future of Telephony.pdf",
here's what I did:
> 
> 1. Installed Fedora 5, and ran "yum update", followed by
"rpm -Uvh kernel
> kernel-devel" (yum would download the i686 version of
"kernel" but the i586
> version of "kernel-devel"). I made sure it had all the
requirements for
> Asterisk (ncurses + ncurses-devel, openssl + openssl-devel, zlib + 
> zlib-devel, and bison)
> 
> 2. Downloaded, unzipped, built, installed the following packages
succesfully:
> asterisk-1.2.13
> asterisk-sounds-1.2.1
> 
> 3. Edited /etc/asterisk/sip.conf thusly:
> [200] ; extension 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
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.

Just to go sure, I use usernames "sip501"...

BR
Anselm

----- Original Message ----- 
From: "Fred" <gkdsh0n02@sneakemail.com>
To: <asterisk-users@lists.digium.com>
Sent: Tuesday, November 14, 2006 12:42 AM
Subject: [asterisk-users] "Username/auth name mismatch" + SIP phone 
can'tconnect?

> Hello
>
> I'm trying to set up Asterisk on an older AMD Duron 700MHz with Fedora
5
> for use with SIP phones and the Linksys 3102 SIP gateway (ie. no FXO card, 
> so no need for zaptel and libpri), but I'm stuck: The GrandStream 
> BudgeTone phone fails registering with Asterisk :-/
>
> Following the "Asterisk - The Future of Telephony.pdf",
here's what I did:
>
> 1. Installed Fedora 5, and ran "yum update", followed by
"rpm -Uvh kernel
> kernel-devel" (yum would download the i686 version of
"kernel" but the
> i586 version of "kernel-devel"). I made sure it had all the
requirements
> for Asterisk (ncurses + ncurses-devel, openssl + openssl-devel, zlib + 
> zlib-devel, and bison)
>
> 2. Downloaded, unzipped, built, installed the following packages 
> succesfully:
> asterisk-1.2.13
> asterisk-sounds-1.2.1
>
> 3. Edited /etc/asterisk/sip.conf thusly:
> [200] ; extension 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
>
> 4. Added an empty section at the bottom of /etc/asterisk/extensions.conf:
>
> [internal]
> ;later
>
> 5. Launched asterisk -vvvvvgc. Launches with no error message
>
> 6. Configured the GrandStream phone with user=200, authname=200, 
> password=test, SIP server=192.168.0.252 (IP of Asterisk server), 
> register=yes
>
> 7. When I plug/unplug the SIP phone, its network icon isn't displayed
(ie.
> there's no connection with the SIP server, and no dial tone), and
here's
> what /var/log/asterisk/messages says:
>
> Nov 13 19:52:33 NOTICE[17922] chan_sip.c: Registration from 
> '<sip:200@192.168.0.252>' failed for '192.168.0.234'
- Username/auth name
> mismatch
>
> => Is this due to wrong settings in sip.conf, the empty section in 
> extensions.conf, the fact that I didn't add the Linksys gateway yet 
> (how?), something else?
>
> Thank you for any tip
> Fred.
>
> _______________________________________________
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.

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.