[asterisk-users] Cannot dial out through SIP provider

[Henrik Woffinden]

Hi,

I'm running Asterisk 1.2.10 bristuffed.
Asterisk is registring perfectly against my provider (musimi.dk), and
incoming calls comes in and are routed fine to either internal  ZAP
(ISDN BRI) and/or SIP.
But....
I can't dial out via SIP (musimi)

sip.conf:
[musimi]
type=friend
host=musimi.dk
username=xxxxxxxx
fromuser=xxxxxxxx
secret=xxxxxxxxxx
domain=musimi.dk
fromdomain=musimi.dk
context=from-sip
;nat=yes
;canreinvite=no
insecure=very
dtmfmode=rfc2833

[9999]
type=friend
context=internal
username=9999
secret=xxxxxxxx
host=dynamic
canreinvite=no
dtfmode=rfc2833
disallow=all
allow=ulaw
callerid="Henrik Woffinden" <9999>
nat=yes
qualify=yes
insecure=very
;mailbox=9999 at from-sip

extensions.conf:
[internal]
;exten => _XXXXXXXX,1,Dial(Zap/g1/${EXTEN},,)
exten => _XXXXXXXX,1,Dial(SIP/${EXTEN}@musimi,,)
exten => _XXXXXXXX,n,Hangup


If I want to dial out via ISDN (Zap which is commented out above), then
it works ok, but via SIP I get the following error message (my own
number is xxxxxxxx and the number I dial is yyyyyyyy - which is a normal
mobile):

-- Registered SIP '9999' at 192.168.9.9 port 29796 expires 3600
-- Executing Dial("SIP/9999-09f2eb28", "SIP/yyyyyyyy at musimi||") in new stack
-- Called yyyyyyyy at musimi
Aug 27 16:38:41 NOTICE[23387]: chan_sip.c:9845 handle_response_invite:
Failed to authenticate on INVITE to '"Henrik Woffinden"
<sip:xxxxxxxx at musimi.dk>;tag=as06ed5480'
-- SIP/musimi-09f34188 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Hangup("SIP/9999-09f2eb28", "") in new stack
== Spawn extension (internal, yyyyyyyy, 2) exited non-zero on
'SIP/9999-09f2eb28'


I hope somebody can tell me what I'm doing wrong here.

-- 
Med venlig hilsen / Best regards,

Henrik Woffinden