[asterisk-users] dial out via fxo gateway

[ACL]

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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