[Asterisk-Users] Asterisk+Sipgate - just one step away..

[Razvan Cosma]

  Hello all,
 I have a working Asterisk setup, also a working sipgate.co.uk account 
(tested with a GrandStream ATA 486), but got stuck in forwarding calls 
from local users to sipgate. Very frustrating, since I feel there's just 
one silly error somewhere.. story follows:
REGISTER both of the local user to * and of the * to sipgate.co.uk is 
successful
but when dialing some random phone number in Linphone in the form 
sip:111111111 at 1.2.3.4 (1.2.3.4 is the * box) I get

    -- Executing SetCallerID("SIP/user-733d", "xxxxxxxx at sipgate.co.uk") 
in new stack
    -- Executing Dial("SIP/user-733d", 
"SIP/1111111111 at sipgate.co.uk|30|tr") in new stack
Outgoing Call for 1111111111111111
111111111111 is not a local user
    -- Called 11111111111 at sipgate.co.uk
Failed to authenticate on INVITE to ''xxxxxxx at sipgate.co.uk" 
<sip:yyyyyyyy at 1.2.3.4>;tag=as319c47a2'
^^^^ this I think is the problem - while the call is redirected, the 
correct number is dialed, Asterisk says it changed the callerid, but 
"yyyyyy" is the local username and "1.2.3.4" is the * address, shouldnt' 
it be ''xxxxxxx at sipgate.co.uk" ?

sip.conf:
[general]
register => xxxxxxxx:ppppppp at sipgate.co.uk/xxxxxx
[sipgate]
type=peer
username=xxxxxxxx
secret=pppppppppp
host=sipgate.co.uk
fromuser=xxxxxxxx
fromdomain=sipgate.co.uk
nat=no
authuser=xxxxxxxx
dtmfmode=info
context=incomingsipgate
context=default
insecure=very
canreinvite=yes
disallow=all
allow=ulaw
allow=alaw

extensions.conf:
[general]
static=yes
writeprotect=yes
[incomingsipgate]
exten => h,1,Hangup
exten => xxxxxxx,1,Dial(SIP/102,20,tr)
[sipgate]
exten => _9.,1,SetCallerID(xxxxxxx at sipgate.co.uk)
exten => _9.,2,Dial(SIP/${EXTEN:1}@sipgate.co.uk,30,tr)
exten => _9.,3,Playback(invalid)
exten => _9.,4,Hangup


Any hints please?
Thank you very much
Razvan