[Asterisk-Users] grandstream handytone 488 fxo

[Soner Tari]

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.

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)

But beware, hangup detection method of HT488 was too simple for my needs. 
Incoming calls may leave the port open indefinetly. (In combination with the 
FXS port of a HT486, it works, but that's it.)

Hope this helps,
Soner

> nope, i havent :\
>
>>> can someone who has a grandstream handytone 488 working with making
>>> outgoing calls through the fxo port please post the parts of their
>>> config that deal with this port? i cant quite seem to get it to make
>>> outgoing calls despite having tried several completely different ways of
>>> making that happen.
>>>
>> I have one but I too haven't been able to make it work.  I've been 
>> looking at the config pages for the 488 and trying to make sense of the 
>> Route to PSTN configuration.  Have you found any documentation for this?