[Asterisk-Users] DTMF is being MUTED by asterisk to/from SIP channels from SIP or ZAP

[Larry Linde]

I am having a problem trying to do inband DTMF passthru via asterisk.

My setup:

PSTN gateway MAXTNT v11.0 SIP (T1 PRI/NT2)
Asterisk HEAD or v1.0 makes no difference (I am using HEAD mostly) 
12/10/04 and 01/17/05 (no difference)
CAC ABII-T100P to/from analog lines to/from asterisk
BTW, I have used a ABI and it works just like the ABII with asterisk.

What I am seeing is:

I make a call from a zap channnel via a CAC ABII -> T100p (zaptel driver 
from same date as asterisk.)
The call goes through just fine. Lets say I call a PSTN phone.
I answer the PSTN I can talk to/from the zap channel like normal.
If I want to send DTMF from the ZAP to the PSTN asterisk mutes the DTMF 
tones even after connect.
I am running G711-ulaw for a codec to/from the PSTN(SIP)
calls both incoming/outgoing to/from asterisk SIP or ZAP work fine for 
voice.
But DTMF will not passthru.

If I bypass asterisk and go direct to the MAXTNT from something like 
kphone. It works fine. DTMF is great.
If I use kphone via asterisk -> PSTN it gets muted.

If I call in from the PSTN to a ZAP channel or a SIP channel via 
asterisk it muted the DTMF tones.

I have tried to switch dtmfmode for the SIP channels to 
inband/rfc2833/info No difference other then
it can break the voicemail decode under asterisk in info.

If I switch asterisk->MAXTNT to RFC2833 I can get DTMF to pass on key 
RELEASE for a short tone. It does
not generate a tone on key press or hold. (Which is what I need {a long 
DTMF tone for a door release})

I must be missing something here, Asterisk can't be that broke that it 
won't pass DTMF correctly.



BTW. I think it works fine if i gate out of asterisk via a T100P (PRI) 
and not a SIP channel.

But I know the SIP gateway is working fine because direct to/from the 
TNT (kphone,ser,cisco,polycom) works.

So its something is asterisk. How do I tell it to get the heck out of 
the way and let my data be free :)



{sip.conf}
[maxtnt-gw]
nat=no
qualify=yes
type=friend
canreinvite=no                  ; Asterisk by default tries to redirect the
host=x.x.x.x
context=maxtnt
reinvite=no
canreinvite=no
disallow=all
allow=ulaw
dtmfmode=inband         ; Choices are inband, rfc2833, or info
accountcode=maxtnt


{zapata.conf}
context=analog
group = 2
signalling=fxo_ks
channel => 1-24

context=testing
callerid="test #"<(612) 111-2222>
channel => 1


{zaptel.conf}
span=1,1,0,esf,b8zs
fxoks=1-24