[Asterisk-Users] Problems with IVR digit recognition

[ismaelg]

  Hello all,

I'm trying setting up an IVR on a Asterisk Soho PBX.

My problem is when I dial the IVR extensión from an Asterisk internal 
extension all goes well, but when I dial the external number of the IVR, 
e.g. 119235656, the PSTN number of my asterisk, I get the same IVR menu 
but when I press on my phone the 1, to select the fist IVR option, or 2, 
to select the second one, (the IVR has only two options),  I can't hear 
anything more on my phone. My phone gets silent. And I lost the rest of 
IVR locution.

I just  add " relaxdtmf=yes"to my first channel on my zapata.conf as I 
read in the forums, without success.

Regards.

Ismael. Gil.

-----------------------Following my 
zapata.conf---------------------------------------------

;
; Zapata telephony interface
;
; Configuration file

[channels]
;
; Default language
;
;language=en
;
; Default context
;
context=default
;
; Switchtype:  Only used for PRI.
;
; national:       National ISDN 2 (default)
; dms100:         Nortel DMS100
; 4ess:           AT&T 4ESS
; 5ess:           Lucent 5ESS
; euroisdn:       EuroISDN
; ni1:            Old National ISDN 1
;
switchtype=national
;
; PRI Dialplan:  Only RARELY used for PRI.
;
; unknown:        Unknown
; private:        Private ISDN
; local:          Local ISDN
; national:       National ISDN
; international:  International ISDN
;
;pridialplan=national
;
; Overlap dialing mode (sending overlap digits)
;
;overlapdial=yes
;
; Signalling method (default is fxs).  Valid values:
; em:      E & M
; em_w:    E & M Wink
; featd:   Feature Group D (The fake, Adtran style, DTMF)
; featdmf: Feature Group D (The real thing, MF (domestic, US))
; featb:   Feature Group B (MF (domestic, US))
; fxs_ls:  FXS (Loop Start)
; fxs_gs:  FXS (Ground Start)
; fxs_ks:  FXS (Kewl Start)
; fxo_ls:  FXO (Loop Start)
; fxo_gs:  FXO (Ground Start)
; fxo_ks:  FXO (Kewl Start)
; pri_cpe: PRI signalling, CPE side
; pri_net: PRI signalling, Network side
; sf:         SF (Inband Tone) Signalling
; sf_w:       SF Wink
; sf_featd:   SF Feature Group D (The fake, Adtran style, DTMF)
; sf_featdmf: SF Feature Group D (The real thing, MF (domestic, US))
; sf_featb:   SF Feature Group B (MF (domestic, US))
; The following are used for Radio interfaces:
; fxs_rx:  Receive audio/COR on an FXS kewlstart interface (FXO at the 
channel bank)
; fxs_tx:  Transmit audio/PTT on an FXS loopstart interface (FXO at the 
channel bank)
; fxo_rx:  Receive audio/COR on an FXO loopstart interface (FXS at the 
channel bank)
; fxo_tx:  Transmit audio/PTT on an FXO groundstart interface (FXS at 
the channel bank)
; em_rx:   Receive audio/COR on an E&M interface (1-way)
; em_tx:   Transmit audio/PTT on an E&M interface (1-way)
; em_txrx: Receive audio/COR AND Transmit audio/PTT on an E&M interface 
(2-way)
; em_rxtx: same as em_txrx (for our dyslexic friends)
; sf_rx:   Receive audio/COR on an SF interface (1-way)
; sf_tx:   Transmit audio/PTT on an SF interface (1-way)
; sf_txrx: Receive audio/COR AND Transmit audio/PTT on an SF interface 
(2-way)
; sf_rxtx: same as sf_txrx (for our dyslexic friends)
;
;signalling=fxo_ls
;signalling=fxs_ls
;########################3Defino los 
canalaes#######################################

signalling=fxs_ks

   callwaiting=yes
   language=en
   context=incoming
   callerid=asreceibed
   relaxdtmf=yes
   channel =>1

signalling=fxo_ks

   callwaiting=yes
   language=en
   context=sales
   callerid=""
   channel =>2

signalling=fxo_ks
   callwaiting=yes
   language=en
   context=reception_bis
   callerid=""
   channel =>3

signalling=fxo_ks
   callwaiting=yes
   language=en
   context=salesi_bis
   callerid=""
   channel =>4

signalling=fxo_ks
   callwaiting=yes
   language=en
   context=salesi_bis2
   callerid=""
   channel =>5

;#####################################################################################
;
; Whether or not to do distinctive ring detection on FXO lines
;
;usedistinctiveringdetection=yes

;
; Whether or not to use caller ID
;
usecallerid=yes
;
; Whether or not to hide outgoing caller ID (Override with *67 or *82)
;
hidecallerid=no
;
; Whether or not to enable call waiting on FXO lines
;
callwaiting=yes
;
; Whether or not restrict outgoing caller ID (will be sent as ANI only, 
not available for the user)
; Mostly use with FXS ports
;
;restrictcid=no
;
; Whether or not use the caller ID presentation for the outgoing call 
that the calling switch is sending
;
usecallingpres=yes
;
; Support Caller*ID on Call Waiting
;
callwaitingcallerid=yes
;
; Support three-way calling
;
threewaycalling=yes
;
; Support flash-hook call transfer (requires three way calling)
;
transfer=yes
;
; Support call forward variable
;
cancallforward=yes
;
; Whether or not to support Call Return (*69)
;
callreturn=yes
;
; Stutter dialtone support: If a mailbox is specified, then when voicemail
; is received in that mailbox, taking the phone off hook will cause
; a stutter dialtone instead of a normal one
;
;mailbox=1234
;
; Enable echo cancellation
; Use either "yes", "no", or a power of two from 32 to 256 if you wish
; to actually set the number of taps of cancellation.
;
echocancel=yes
;
; Generally, it is not necessary (and in fact undesirable) to echo cancel
; when the circuit path is entirely TDM.  You may, however, reverse this
; behavior by enabling the echo cancel during pure TDM bridging below.
;
Echocancelwhenbridged=yes
;
; In some cases, the echo canceller doesn't train quickly enough and there
; is echo at the beginning of the call.  Enabling echo training will cause
; asterisk to briefly mute the channel, send an impulse, and use the impulse
; response to pre-train the echo canceller so it can start out with a much
; closer idea of the actual echo.
;
;echotraining=yes
;
; If you are having trouble with DTMF detection, you can relax the
; DTMF detection parameters.  Relaxing them may make the DTMF detector
; more likely to have "talkoff" where DTMF is detected when it
; shouldn't be.
;
;relaxdtmf=yes
;relaxdtmf=yes

;
; You may also set the default receive and transmit gains (in dB)
;
rxgain=0.0
txgain=0.0
;
; Logical groups can be assigned to allow outgoing rollover.  Groups
; range from 0 to 31, and multiple groups can be specified.
;
group=1
;
; Ring groups (a.k.a. call groups) and pickup groups.  If a phone is ringing
; and it is a member of a group which is one of your pickup groups, then
; you can answer it by picking up and dialing *8#.  For simple offices, just
; make these both the same
;
callgroup=1
pickupgroup=1

;
; Specify whether the channel should be answered immediately or
; if the simple switch should provide dialtone, read digits, etc.
;
immediate=no
;
; CallerID can be set to "asreceived" or a specific number
; if you want to override it.  Note that "asreceived" only
; applies to trunk interfaces.
;
;callerid=2564286000
;
; AMA flags affects the recording of Call Detail Records.  If specified
; it may be 'default', 'omit', 'billing', or 'documentation'.
;
;amaflags=default
;
; Channels may be associated with an account code to ease
; billing
;
;accountcode=lss0101
;
; ADSI (Analog Display Services Interface) can be enabled on a per-channel
; basis if you have (or may have) ADSI compatible CPE equipment
;
;adsi=yes
;
; On trunk interfaces (FXS) and E&M interfaces (E&M, Wink, Feature Group D
; etc, it can be useful to perform busy detection either in an effort to
; detect hangup or for detecting busies
;
;busydetect=yes
;
; If busydetect is enabled, is also possible to specify how many
; busy tones to wait before hanging up. The default is 4, but
; better results can be achieved if set to 6 or even 8. Mind that
; higher the number, more time is needed to hangup a channel, but
; lower is probability to get random hangups
;
;busycount=4
;
; On trunk interfaces (FXS) it can be useful to attempt to follow the 
progress
; of a call through RINGING, BUSY, and ANSWERING.   If turned on, call
; progress attempts to determine answer, busy, and ringing on phone lines.
; This feature is HIGHLY EXPERIMENTAL and can easily detect false answers,
; so don't count on it being very accurate.  Also, it is ONLY configured for
; standard U.S. tones.  This feature can also easily detect false hangups.
; The symptoms of this is being disconnected in the middle of a call for no
; reason.
;
;callprogress=yes
;
; Select which class of music to use for music on hold.  If not specified
; then the default will be used.
;
;musiconhold=default
;
; PRI channels can have an idle extension and a minunused number.  So long
; as at least "minunused" channels are idle, chan_zap will try to call
; "idledial" on them, and then dump them into the PBX in the "idleext"
; extension (which is of the form exten at context).  When channels are needed
; the "idle" calls are disconnected (so long as there are at least "minidle"
; calls still running, of course) to make more channels available.  The
; primary use of this is to create a dynamic service, where idle channels
; are bundled through multilink PPP, thus more efficiently utilizing
; combined voice/data services than conventional fixed mappings/muxings.
;
;idledial=6999
;idleext=6999 at dialout
;minunused=2
;minidle=1
;
; Configure jitter buffers in zapata (each one is 20ms, default is 4)
;
;jitterbuffers=4
;
; Each channel consists of the channel number or range.  It
; inherits the parameters that were specified above its declaration
;
;callerid="Green Phone"<(256) 428-6121>
;channel => 1
;callerid="Black Phone"<(256) 428-6122>
;channel => 2
;callerid="CallerID Phone" <(256) 428-6123>
;callerid="CallerID Phone" <(630) 372-1564>
;callerid="CallerID Phone" <(256) 704-4666>
;channel => 3
;callerid="Pac Tel Phone" <(256) 428-6124>
;channel => 4
;callerid="Uniden Dead" <(256) 428-6125>
;channel => 5
;callerid="Cortelco 2500" <(256) 428-6126>
;channel => 6
;callerid="Main TA 750" <(256) 428-6127>
;channel => 44
;
; For example, maybe we have some other channels
; which start out in a different context and use
; E & M signalling instead.
;
;context=remote
;sigalling=em
;channel => 15
;channel => 16

;signalling=em_w
;
; All those in group 0 I'll use for outgoing calls
;
; Strip most significant digit (9) before sending
;
;stripmsd=1
;callerid=asreceived
;group=0
;signalling=fxs_ls
;channel => 45

;signalling=fxo_ls
;group=1
;callerid="Joe Schmoe" <(256) 428-6131>
;channel => 25
;callerid="Megan May" <(256) 428-6132>
;channel => 26
;callerid="Suzy Queue" <(256) 428-6233>
;channel => 27
;callerid="Larry Moe" <(256) 428-6234>
;channel => 28
;
; Sample PRI (CPE) config:  Specify the switchtype, the signalling as
; either pri_cpe or pri_net for CPE or Network termination, and generally
; you will want to create a single "group" for all channels of the PRI.
;
; switchtype = national
; signalling = pri_cpe
; group = 2
; channel => 1-23

;
;  Used for distintive ring support for x100p.
;  You can see the dringX patterns is to set any one of the 
dringXcontext fields
;  and they will be printed on the console when an inbound call comes in.
;
;dring1=95,0,0
;dring1context=internal1
;dring2=325,95,0
;dring2context=internal2
; If no pattern is matched here is where we go.
;context=default
;channel => 1

;#######################333Pro0bando conexiópn con el 
TRunk-telefonoca#######

signalling=fxs_ks ; X100P
echocancel=yes ; You can set this to 32, 64, or 128, tweak to your needs.
echocancelwhenbridged=yes
echotraining=400 ; Asterisk trains to the beginning of the call, number 
is in milliseconds
callerid=asreceived
;group=1
context=default ; Points to the default context of your extensions.conf
channel => 1

;#######################################################################3