[Asterisk-Users] grandstream handytone 488 fxo

[Dave Cotton]

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>