[asterisk-users] "dtmf transcoding" with asterisk

[Hagai Sela (TA)]

Hi,
I am trying to configure asterisk to translate between rfc2833 and
inband DTMF.
I have a cisco gateway which is configured as a trunk, and a cisco IP
phone which is registered to asterisk. The gateway does not support
rfc2833 and the IP phone does. 
I tried changing directrtpsetup to "no", and that didn't help. I tried
changing "canreinvite" to "no", but that didn't help either.
 
I tried adding some device-specific configuaration to sip.conf, and now
my calls are rejected with a status code of "404 not found".
This is what I added in sip.conf:
 
[6102]

type=friend

canreinvite=no

host=dynamic

dtmfmode=rfc2833

[trunk_1]

type=peer

host=192.168.20.58

canreinvite=no

dtmfmode=inband

What am I doing wrong?

Hagai.

 
 
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