DTMF and G729

Asterisk Asterisk Support
1

I recently added G729 licenses to one of our Asterisk servers. When a call connects with G729 , DTMF ceases to be recognized by the Asterisk system.
I have tried auto and rfc2833 for dtmfmode. Here is a copy of the sip.conf

[general]
port = 5060           ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
disallow=all
allow=g729
allow=ulaw
allow=alaw
relaxdtmf=yes
dtmfmode=auto
;dtmfmode=rfc2833
canreinvite=no
qualify=yes
jbenable=yes
t38pt_udptl=yes ; Default false
context=from-trunk ; Send unknown SIP callers to this context
tos_sip=cs3
tos_audio=ef
progressinband=never
subscribecontext=ext-local
limitonpeers = yes

I have tried searching google and this forum for possible answers. Has anyone else using G729 encountered this issue and if so , could you point me in the right direction to resolve this?

2

CODEC should have no bearing on rfc2833 (or any off-band mode). Not quite sure how auto would behave but what messages does CLI display when rfc2833 is used? Is the UA configured to use rfc2833?

3

I believe you need to use dtmfmode=rfc2833 so DTFM information is “extracted” and send in parallel with media stream.

If inband DTMF is used it means DTFM tones are send as any other sound or voice in the media stream so the other side has to interpret it. In this case you can not use any codec except for ulaw/alaw because all narrow-band codecs distort sound in some way. Which makes inband DTFM not understandable by the other side.