[asterisk-users] setting dtmf mode for a particular peer

[Brent Davidson]

Brian J. Murrell wrote:
> On Wed, 2008-04-09 at 17:21 -0500, Brent Davidson wrote:
>   
>
> Does anyone know if Asterisk will convert an inband DTMF from one sip
> channel to an info or rfc2833 DTMF on another (i.e. bridged) SIP
> channel?
You might also try "canreinvite=no" for both your phone and the sip 
peer.  I think it's normal procedure for Asterisk to drop out of the 
call path once the call is established between two peers.  The 
canreinvite directive forces asterisk to remain as an intermediary, and 
it will probably do the transcoding that way.  If I'm not mistaken this 
is also useful for making calls between two system that have no common 
codecs.

-Brent