[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)
>>
> 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?

That may be related with the dtmfmode. Can you try inband? I believe rfc2833 
should work too, but once you have it working with inband, you can test the 
rest.

Also I think you'd like to use PCMU codec on HT488, other codecs may cause 
DTMF detection problems (iLBC seems fine though).

In short, I would play with DTMF and codec parameters on both sides.
Hope this helps,
Soner