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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