asterisk users - Sep 2008 - dial out via fxo gateway

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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On Thu, Sep 4, 2008 at 4:44 PM, ACL <access_you at yahoo.com>
wrote:
> 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.
>
I usually have a simple outbound context

[outbound]
exten => _9X.,1,Dial(Zap/g0/${EXTEN:1})
exten => _9X.,n,Congestion()
exten => _9X.,n,Hangup()

Be warned that the above dialplan will allow calls with anykind of
numbers (even international). So be sure to pattern match depending of
where the calls should go.

Don't forget to include the [outbound] context in whatever context
your SIP extention is in.

Steve