Using same extension number for outgoing and incoming both internal and PSTN

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Asterisk Users 1 Comment

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

One thought on - Using same extension number for outgoing and incoming both internal and PSTN

  • 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
    @ 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