[asterisk-users] [Asterisk-Users]Using same extension number for outgoing and incoming both internal and PSTN

[Samuel Sappa]

Sorry if this question already asked.
I'm implementing Voip with asterisk and grandstream gxw4108, according
from the manual, for connecting with PSTN I must configure one SIP
account and assign that for dialing the PSTN so in my sip.conf I
configure SIP account(extension) :

[1401]
type=friend
username=1401
secret=1401
host=dynamic
context=my-office
insecure=port

in my extension.con
[my-office]
exten=>1401,1,Dial(SIP/1401,60)
exten=>_NXXNXXXX,1,Dial(SIP/${EXTEN}@1401)

but the problem is when I dial the number for the PSTN it's run/dial
on internal extension, from the asterisk guru website it's wrote to
separate the incoming and out going
in sip.conf
[1401]
type=friend
username=1401
secret=1401
host=dynamic
context=my-office-in
insecure=port

[1401]
type=friend
username=1401
secret=1401
host=dynamic
context=my-office-out
insecure=port

in extension.conf
[my-office-in]
exten=>1401,1,Dial(SIP/1001,60)
[my-office-out]
exten=>_NXXNXXXX,1,Dial(SIP/${EXTEN}@1401)

but still with this won't work too
My question it's
Is it my configuration true/correct or if there any other way for my problem
I'm using 1 Stage Dialing and the asterisk server and Grandstream
using different IP Address 192.168.101.xxx (for asterisk server) and
192.168.14.xxx (for grandstream gateway)
thank you for helping
-- 
Regards
Samuel Sappa,