[Asterisk-Users] grandstream handytone 488 fxo

[Soner Tari]

Of course... Those are the basics to get HT488 working for the OP.  In this 
thread I am not trying to show how to create dialplans.

> On Tue, 2005-08-30 at 17:11 +0300, Soner Tari wrote:
>> I use HT488, and I can make and receive FXO calls. It's actually quite
>> simple, you create a SIP acount in sip.conf. On the FXO section of HT488 
>> web
>> admin page you enter these registration values. When you reboot the HT488
>> you should see it registering on Asterisk CLI.
>>
>> What's left is a dialplan line in extensions.conf like this:
>> exten => 9,1,Dial(SIP/<sip acount name>,10)
>>
>> That's for making outbound calls.
>
> This means that you have 2 stage dialing, 9 gives you an outside dial
> tone. Won't it work with single stage?
>
> _9.,1,Dial(${DIALOUTPSTN}/${EXTEN:1})
>
>
>> Once you've done this, you can direct incoming calls to a context like 
>> this:
>> exten => 50,1,Goto(MainMenu,s,1)
>>
>> You should enter 50 to "Forward to VoIP" box at the bottom of HT488 
>> config
>> page also. (Choose an extension as you like instead of 50)
>
> Problem with this is no CallerID it'll always be 50.
>
>
> -- 
> Dave Cotton <dcotton at linuxautrement.com>
>
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