[Asterisk-Users] DTMF Talk off

[John Millican]

Hello all,
I have seen some chatter again about DTMF.  I see most of the talk about DTMF 
around not being able to get an external IVR to recognize digits, not a big 
issue for me at this time but sill interesting.  My issue though, is with 
talk off on a zap channel.  It seems to be getting worse or maybe my patience 
is thinning.  All my calls go out and come in pstn through an FXO as I do not 
have high speed available here at home.  My Current setup is:

Phone-->PAP2--> * --->PSTN--->Voip number to * at another location(that has 
high speed)--->send to VoIP provider

I read a post about talked about the length of the DTMFish sound.  I also 
remeber seing something about relaxdtmf being set to something other than yes 
or no, so I looked in chan_zap.c and found  relaxdtmf in many places but it 
looked to my inexperienced eye that it could only be set to 'yes' or 'no', 
but i did find some mention of tonlength (at line 10858) 
followed that to zaptel.c (line 3357) where it said :
if ((tdp.dtmf_tonelen > 4000 ) || (tdp.dtmf_tonelen < 10 ))
    return -EINVAL
Which I am guessing means unless the dtmf is between these 2 values ignore it.
Any ideas what might happen if i increased the minimum time value? if this is 
indeed what this is referring to?


Zapata.conf:
[channels]
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
busydetect=yes
busycount=6
echocancel=128
echocancelwhenbridged=yes
echotraining=yes
rxgain=0
txgain=0
immediate=no
context=default
signalling=fxs_ks
channel => 1
<same for channel 2>

zaptel.conf:
loadzone = us
fxsks=1
fxsks=2

extensions.conf:
exten => s,1,  NoOp(${CALLERID} time ${DATETIME});
exten => s,2,  Dial(sip/677&sip/666,30,tT);
exten => <a bunch of stuff to do with agi look ups and voicemail 
leave/retrieve>

All very basic and works like a charm except for the talk off.
Thanks in advance to all,
John M