[Asterisk-Users] grandstream handytone 488 fxo

[Casey Boone]

i greatly appreciate the information and will be giving it a whirl later 
today :)

Casey

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.
> 
> 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? 
> 
> 
> _______________________________________________
> --Bandwidth and Colocation sponsored by Easynews.com --
> 
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
> 
>