[asterisk-users] "Username/auth name mismatch" + SIP phone can'tconnect?

[Dovid B]

----- Original Message ----- 
From: "Fred" <gkdsh0n02 at sneakemail.com>
To: <asterisk-users at 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 at 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.