sip proxy

hello,
i want to connecte my astersk server with un @ of sip proxy ,
i make
and this
in extensions.conf i have make this
in ade fisk i have the account 1003
and when i dial a number i have this message
i want to connecte my astersk server with un @ of sip proxy ,
i make
in sip.confregister=> 112233:password@"@sip server"
and this
[112233]
type=peer
username=112233
fromuser=112233
secret=password
context=incoming-SIP-context-in-extensions.conf
host=@sip server
[1003]
secret=mypassword
username=1003
host=dynamic
auth-plaintext
dtmfmode=inband
context=outgoing-SIP-context-in-extensions.conf
type=friend
in extensions.conf i have make this
[outgoing-SIP-context-in-extensions.conf]
exten=>_00XXXXXXXXXXX,1,Dial(SIP/112233:password@"@sipserver"/${EXTEN})
[incoming-SIP-context-in-extensions.conf]
exten=>_112233,1,Dial(SIP/1003)
in ade fisk i have the account 1003
and when i dial a number i have this message
-- Executing Dial("SIP/1003-0817b880", "SIP/112233:password@"@sip server"/number what i have called") in new stack
May 23 14:44:12 WARNING[14777]: chan_sip.c:1991 create_addr: No such host: @"@sip server"/number what i have calledMay 23 14:44:12 WARNING[14777]: chan_sip.c:1991 create_addr: No such host: @"@sip server"/number what i have called
May 23 14:44:12 NOTICE[14777]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
May 23 14:44:12 NOTICE[14777]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'SIP/1003-0817b880' status is 'CHANUNAVAIL'
is this configuration correct ???
if not can you help me to configure sip.conf and extensions.conf corectelly!!