Get trouble when connect 2 asterisks through sip trunk!

Asterisk Asterisk Support
1

Hi everyone!

I try to connect 2 asterisks through sip trunk, with my configuration as follow:

Trixbox 1 (192.168.1.200)

With sip.conf

[general]
register => toronto:welcome@192.168.1.212

[osaka]
type=friend
secret=welcome
context=osaka_incoming
host=dynamic
disallow=all
allow=ulaw

[2000]
type=friend
host=dynamic
context=phones

[2001]
type=friend
host=dynamic
context=phones

With extensions.conf

[globals]

[general]
autofallthrough=yes

[incoming_calls]

[phones]
include => internal
include => remote

[internal]
exten => _2XXX,1,NoOp()
exten => _2XXX,n,Dial(SIP/${EXTEN},30)
exten => _2XXX,n,Playback(the-party-you-are-calling&is-curntly-unavail)
exten => _2XXX,n,Hangup()

[remote]
exten => _1XXX,1,NoOp()
exten => _1XXX,n,Dial(SIP/osaka/${EXTEN})
exten => _1XXX,n,Hangup()

[osaka_incoming]
include => internal

Trixbox 2 (192.168.1.212)

With sip.conf

[general]
register => osaka:welcome@192.168.1.200

[toronto]
type=friend
secret=welcome
context=toronto_incoming
host=dynamic
disallow=all
allow=ulaw

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

[1001]
type=friend
host=dynamic
context=phones

With extensions.conf

[globals]

[general]
autofallthrough=yes

[incoming_calls]

[phones]
include => internal
include => remote

[internal]
exten => _1XXX,1,NoOp()
exten => _1XXX,n,Dial(SIP/${EXTEN},30)
exten => _1XXX,n,Playback(the-party-you-are-calling&is-curntly-unavail)
exten => _1XXX,n,Hangup()

[remote]
exten => _2XXX,1,NoOp()
exten => _2XXX,n,Dial(SIP/toronto/${EXTEN})
exten => _2XXX,n,Hangup()

[toronto_incoming]
include => internal

;-------------------------
When I register 2 sip phones: one as 1000 and the other as 2000. I can’t call between them and the error is “403 Fobidden”. Please let me know the reason !
Thank a lot !