[Asterisk-Users] grandstream handytone 488 fxo

[Dave Cotton]

On Wed, 2005-08-31 at 09:54 -0300, Keith Yoder wrote:
> Soner Tari escreveu:
> 
> > 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)
> >
> I've tried your example shown here.  When I dial 9 I get dial tone from 
> the HT488 but I when I try to dial, nothing happens (i.e. I keep hearing 
> dial tone even though I'm dialing).  Any ideas?

What are your DTMF settings?

I had all sorts of weird problems with a differant manufacturers ATA
because of this.


-- 
Dave Cotton <dcotton at linuxautrement.com>