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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