[asterisk-users] Problem with "Background" DTMF detection with A200D

[Alvin Austin]

Hi all,

I'm having trouble with Background DTMF detection, and would appreciate 
any suggestions.

A call comes in to a Sangoma A200D PSTN line.  A standard menu welcome 
is used.  Most of the time, callers have to wait until the message 
completes in order to have their selection recognized.  People end up 
having to press the option number several times. Occasionally, you can 
press the desired option digit during the message and it will be 
selected right away while the Background message is still playing (this 
is what I want all the time).  Any suggestions?

Environment: Asterisk 1.2.10, zaptel-1.2.7, wanpipe-beta7-2.3.4.tgz
Machine has lots of horsepower: Pentium D 3.2 GHz, 2 GB RAM,

[general]
priorityjumping=no
autofallthrough=no
(...)

[from-pstn]
; Inbound calls from PSTN line
exten => s,1,NoOp(TIMESTAMP: ${TIMESTAMP})
exten => s,2,NoOp(CONTEXT: ${CONTEXT})
exten => s,3,NoOp(CALLERIDNUM: ${CALLERIDNUM})
exten => s,4,NoOp(CALLERIDNAME: ${CALLERIDNAME})
exten => s,n,Goto(mainmenu,s,1)

[mainmenu]
exten => s,1,NoOp(Main Menu)
exten => s,n,Wait,1
exten => s,n,Answer
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Playback(silence-1sec)
exten => s,n,Playback(silence-1sec)

exten => s,n,Background(mainmenu)
;   Thank you for calling xxx.
;    Please press 1 for AA;
;    2 for BB;
;    3 for CC;
;    or 4 for DD.
;    Press 0, or stay on the line for reception.

exten => 1,1,NoOp(Menu 1 - Dialing SIP/101 AA)
exten => 1,n,Dial(SIP/101,20,t)
exten => 1,n,Playback(silence-1sec)
exten => 1,n,Voicemail(u101)
exten => 1,n,Hangup

exten => 2,1,NoOp(Menu 2 - Dialing SIP/102 BB)
exten => 2,n,Dial(SIP/102,20,t)
exten => 2,n,Playback(silence-1sec)
exten => 2,n,Voicemail(u102)
exten => 2,n,Hangup

exten => 3,1,NoOp(Menu 1 - Dialing SIP/103 CC)
exten => 3,n,Dial(SIP/103,20,t)
exten => 3,n,Playback(silence-1sec)
exten => 3,n,Voicemail(u103)
exten => 3,n,Hangup

exten => 4,1,NoOp(Menu 1 - Dialing SIP/104 DD)
exten => 4,n,Dial(SIP/104,20,t)
exten => 4,n,Playback(silence-1sec)
exten => 4,n,Voicemail(u104)
exten => 4,n,Hangup

exten => 0,1,NoOp(Menu 0 - Dialing SIP/100)
exten => 0,n,Dial(SIP/100,20,t)
exten => 0,n,Playback(silence-1sec)
exten => 0,n,Voicemail(u100)
exten => 0,n,Hangup

exten => #,1,NoOp(Menu # - Access VOICEMAIL)
exten => #,n,Playback(silence-1sec)
exten => #,n,VoiceMailMain()
exten => #,n,Hangup
;
exten => t,1,NoOp(Menu t - Goto mainmenu,0,1)
exten => t,n,Goto(mainmenu,0,1)
;
exten => i,1,NoOp(Menu i - Playback pbx-invalid)
exten => i,n,Playback(pbx-invalid)
exten => i,n,Goto(mainmenu,s,1)

;end of [mainmenu]
;------------------------


In the zapata.conf file, the relevant parts are:
[trunkgroups]

[channels]
language=en
context=default
rxwink=300 ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
cidsignalling=bell
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
musiconhold=default
echocancel=yes
echocancelwhenbridged=yes
rxgain=3.0
txgain=0.0
immediate=no
faxdetect=no

group=1
signalling=fxs_ks
context=from-pstn
language=en
channel => 1

group=1
signalling=fxs_ks
context=from-pstn
language=en
channel => 2

group=1
signalling=fxs_ks
context=from-pstn
language=en
channel => 3

group=1
signalling=fxs_ks
context=from-pstn
language=en
channel => 4

group=1
signalling=fxs_ks
context=from-pstn
language=en
channel => 5

group=1
signalling=fxs_ks
context=from-pstn
language=en
channel => 6

;-----------------------


Thanks for any ideas,
Alvin