ITSP out calls problem; whhat is wrong?

1.Asterisk is installed on Linux(firewall) with two net interfaces
ext 213.210.156.21
int 192.168.1.161
localclients - softphones(sjphone)
2. and connected to ITSP sipnet.ru (see sip.conf)

Problem:
When I call from localnet dialing sipnet.ru_number(for examp. 2009991) - OK, but when i call outside number 7xxxxxxxxxx I get a message
"call rejected: 503 service unavailable"

I configured other ip-pbx(3cx) without problem, but here I have some diffuculty…

Below sip.conf and extensions.conf

sipnet.conf

[general]
register => 3485125:mysipnetrupsw:mysipnetrulogin@sipnet.ru/sipnetru
context=default
allowoverlap=yes
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
externip=213.156.210.21
localnet=192.168.1.0/24
recordhistory=yes
canreinvite=yes
relaxdtmf = no
dtmfmode = auto
disallow=all
allow=alaw
allow=ulaw
allow=gsm

[authentication]

[1000]
type=friend
host=dynamic
context=phones

[200]
type=friend
host=dynamic
secret=200
regexten=200
context=phones

[206]
type=friend
host=dynamic
secret=206
context=phones

[207]
type=friend
username=207tst
host=dynamic
secret=207
context=phones

[sipnetru]
type=peer
host=sipnet.ru
fromuser=mysipnetrulogin
username=3485125
secret=mysipnetrupsw
dtmfmode = auto
disallow=all
allow=g729a
allow=alaw
outboundproxy=sipnet.ru
regexten=1000
canreinvite=no
context=outgoing_calls

extensions.conf

[globals]

[general]
;autofallthrough=yes

[default]
exten => s,1,Verbose(1|Unrouted call handler)
exten => s,n,Answer
exten => s,n,Wait(1)
exten => s,n,Playback(tt-weasels)
exten => s,n,Hangup()

[incoming_calls]
exten => _X.,1,NoOp()
exten => _X.,n,Dial(SIP/1000)

[outgoing_calls]

exten => _X.,1,Dial(SIP/sipnetru/${EXTEN})

[internal]

exten => 1000,1,Dial(SIP/200,30)

exten => 500,1,Playback(demo-abouttotry); Let them know what’s going on
exten => 500,n,Dial(IAX2/guest@pbx.digium.com/s@default) ; Call the Asterisk demo
exten => 500,n,Playback(demo-nogo) ; Couldn’t connect to the demo site
exten => 500,n,Goto(s,6) ; Return to the start over message.

exten => 600,1,Playback(demo-echotest) ; Let them know what’s going on
exten => 600,n,Echo ; Do the echo test
exten => 600,n,Playback(demo-echodone) ; Let them know it’s over
exten => 600,n,Goto(s,6) ; Start over

exten => 200,1,Dial(SIP/200)
exten => 206,1,Dial(SIP/206)
exten => 207,1,Dial(SIP/207)

[phones]
include => internal
include => outgoing_calls

I find solution on www.asterisk-support.ru.