[Asterisk-Users] Limit DTMF tones

[Andrew Thompson]

Flynn wrote:
  > Possibly, but his working configuration most likely doesn't use SIP (I
> would presume):
> 
> It has the Digium 2 FXO/ 2 FXS card in it.  I have two Lines brought in
> to the fxo ports and 2 standard 2500 analog sets for the prisoners to
> use to dial out.

Yeah, I saw that, but the replies I'd seen so far were not looking real 
promising, so I thought I'd throw out another idea.

Even if the handsets were ruggedized, a Sipura could sit in between them 
and asterisk.

Critchfield's response about the bridge code seems the place to look, 
but that's going to require coding and testing.

If a SIP adapter could be dropped in and as a side effect of the 
configuration it broke sending DTMF out, only a few changes to the 
dialplan would be required to get things back in order.

Anyway, it was just an idea, and he did say he was looking for ideas.


-- 
Andrew Thompson
http://aktzero.com/