Why is DTMF not processed correctly?

Question

Asterisk

Greetings. There is a SIP trunk from RT (the so-called new telephony) which falls on IVR with the possibility of additional dialing.
Apparently, DTMF is not processed correctly and the call sometimes goes to a random Extention instead of the desired one.
Here are the trunk settings:
username=user1
type=friend
secret=111111
registersip=yes
qualify=yes
nat=yes
insecure=port,invite
host=111111.15.rt.ru
fromuser=user1
fromdomain=111111.15.rt.ru
dtmfmode=rfc2833
disallow=all
context=from-trunk
canreinvite=no
allow=alaw&ulaw&gsm
"dtmfmode=" did auto, inband - no difference.
Thank you.