dial out via fxo gateway

[ACL <access_you@xxxxxxxxx>]

My current config:

pstn -> audiocodes fxo gateway -> asterisk -> xlite

every fxo ports are registered with asterisk

I have this extensions.conf

exten => 111,1,answer
exten => 111,n,dial(sip/fxo1)
exten => 111,n,hangup

If we dial 111 by xlite, I could hear pstn dialing tone. I could key in a phone no and connect to the called party. this is a two stage dialing.

How could we preset a phone no. in the extensions.conf without having the sip client keys in the phone no (ONE STAGE DIALING)? I do not want to preset the phone no. in fxo gateway.  the phone no. must be modifiable.

pls kindly advise.

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