asterisk users - Sep 2011 - Using same extension number for outgoing and incoming both internal and PSTN

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,

Hey,

I don;t think asterisk-guru could've been wrong on this one - possibly
different scenario than your's. Anyway I see what you did there ! There is
no need for separate context for  incoming or outgoing if you don't want.
What you are doing is *exten=>_NXXNXXXX,1,Dial(SIP/${EXTEN}@1401**) *
*
*
When you defined the SIp user/peer [1401] you stated context for handling
dial request as "my-office" and when you tried dialling out you told
asterisk to dial the requested number located at 1401 which should've been
@<IP.OF.Grandstream.GW> if calls need to be dialed to gateway and If your
gateway just accepts SIP based (w/o auth) calls.

*exten=>_NXXNXXXX,1,Dial(SIP/${EXTEN}@192.168.14.???**)  *
*
*
If your gateway shows attitude in serving direct request you may need to
create user in gateway and telling asterisk to register on Grandstream as a
user and dial-out using that user like.

*exten=>_NXXNXXXX,1,Dial(SIP/${EXTEN}@gstream-user**)  *
*
*
There could be more possible alternatives to successfully dial-out using one
context for handling incoming an out going/ preferred is you create separate
contexts.

Regards,
- Sammy

On Tue, Sep 20, 2011 at 8:13 PM, Samuel Sappa <cihuy916 at gmail.com>
wrote:
> 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,
>
> --
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