[Asterisk-Users] Passing DTMF Transparently

[Rich Adamson]

> > 	Does anyone know if it is possible to setup asterisk such that
> > it passes DTMF Tones through from One channel to the next transparently.
> 
> I don't believe this is possible, no. If you are using all Zap channels 
> (TDM cards) and don't enable _any_ DTMF-controlled features in the 
> Dial() application, it might work this way, but not intentionally 
> (meaning it could change in the future) :-)

I'm not the OP, but based on 20+ years of detailed engineering 
experience for a large US telco, the defacto stardard for all analog 
telephony gear is to pass dtmf inband once a call is considered 
answered. 

Given the OP's stated objective, there would not appear to be any 
realistic way to accomplish his objective without resorting to 100% 
inband beginning with the originating phone (sip or otherwise), which 
he is not going to be able to accomplish with sip phones while also 
maintaining other compatibilities.

Also, since there are a large number of commercial pbx's that use 
electronic phones and pbx-controllled dtmf generation (whose timing is 
not under user control), whatever design the OP is looking for won't 
work with a large number of production pbx's. Therefore, the OP's design
needs to change as opposed to modifying * to accomplish the objective.