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

[Anselm Martin Hoffmeister]

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