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.
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
