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 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)?
pls kindly advise.