DTMF Übertragung über ZAP Channel

[thomhor]

Hallo!
Ich habe folgendes Problem:Asterisk erkennt von der ISDN Seite her keine DTMF Sequenzen. Mein Systemaufbau:

Analog TLN -->
System TLN --> HICOM 118E --> HFC-Karte (TE) - Asterisk (1.2)
PSTN -->

Auch von analogen Endgeräten aus werden keine DTMF Töne erkannt.
Muss ich in der zaptel/ zapata.conf noch etwas ändern um Asterisk Inband auf DTMF hören zu lassen?

Thomas

 

[NetRacer]

Moin

ich habe ein ähnliches Problem. Bei allen Gesprächen über extern ISDN funktioniert DTMF nicht. CLI zeigt folgende Meldung, wenn ich eine Tasta drücke:

[Sep 10 15:49:37] WARNING[10285]: chan_zap.c:1229 zt_digit_begin: Couldn't dial digit 4

Meine Configdateien sind folgend:

zaptel.conf

defaultzone=de
loadzone=de

span=1,1,3,ccs,ami
bchan=1-2
dchan=3

span=2,1,3,ccs,ami
bchan=4-5
dchan=6

zapata.conf:

[channels]
;
;defaults
;language=de   ; you'll need to have German voiceprompts installed for this
switchtype = euroisdn
signalling = bri_cpe_ptmp   ; this is TE mode! Use bri_net_ptmp for standard NT mode
pridialplan=local
echocancel=yes
;immediate=yes   ; is this correct, isn't this only for analog handsets?
;setcallerid(""<${CALLERIDNUM}>)  ; is this correct, shouldn't it rather be "callerid=..."?
;callerid=asreceived
;usecallerid=yes
;context=default
usecallingpres=yes
immediate=no
overlapdial=yes
;toneduration=600

;  card 1
group = 1
context=extern238
channel => 1-2

;  card 2
signalling = bri_cpe_ptmp
group = 2
context=extern410
channel => 4-5

sip.conf:

[general]
context=default                 ; Default context for incoming calls
bindport=5060                   ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
allow=ulaw                      ; Allow codecs in order of preference

dtmfmode = rfc2833          
allowsubscribe=yes
notifyringing=yes
notifyhold=yes
useclientcode=yes
limitonpeers = yes



[5555]
type=friend
secret=5555
;qualify=1000                   ; Qualify peer is no more than 200ms away
nat=yes
host=dynamic                    ; This device registers with us
;username=goran                 ; Username to use when calling this device before registration
context=interne-telefone
mailbox=5555
subscribecontext=interne-telefone
call-limit=4


[4444]
type=friend
secret=4444
;qualify=1000                   ; Qualify peer is no more than 200ms away
nat=yes
host=dynamic                    ; This device registers with us
;username=goran                 ; Username to use when calling this device before registration
context=interne-telefone
subscribecontext=interne-telefone
call-limit=4

[3333]
type=friend
secret=3333
nat=yes
host=dynamic
context=interne-telefone
subscribecontext=interne-telefone
call-limit=4

[7777]
type=friend
secret=7777
nat=yes
host=dynamic
context=interne-telefone
subscribecontext=interne-telefone

[8888]
type=friend
secret=8888
nat=yes
host=dynamic
context=interne-telefone
subscribecontext=interne-telefone

[9999]
type=friend
secret=9999
nat=yes
host=dynamic
context=interne-telefone
subscribecontext=interne-telefone

[general]
autofallthrough=no

[interne-telefone]

exten => 100,1,Ringing
exten => 100,n,Wait(3)
exten => 100,n,MeetMe(1234,Mic) ; 
exten => 100,n,Hangup


exten => 5555,1,Dial(SIP/5555, 15)
exten => 5555,2,VoiceMail(5555)

exten => 6666,1,Dial(SIP/6666)
exten => 4444,1,Dial(SIP/4444)
exten => 3333,1,Dial(SIP/3333)

exten => 7777,1,Dial(SIP/7777)
exten => 8888,1,Dial(SIP/8888)
exten => 9999,1,Dial(SIP/9999)



; W�hlen

exten => _0XX.,1,set(CALLERID(num)=238238238)
;exten => _0XX.,1,SetCallerPres(prohib)
; Matching von den individuellen Teilnehmern
;exten => _0XX./5555,1,set(CALLERID(num)=328328328)
;exten => _0XX./3333,1,SetCallerPres(prohib)


exten => _0XX.,n,Set(SPYGROUP=10005)

exten => _0XX.,n,Dial(ZAP/g1/${EXTEN:1}) ; ISDN out

exten => 123,1,ChanSpy(,g(10005))
exten => 123,n,Hangup()

; mailbox defaukt for snom defaults
exten => asterisk,1,VoicemailMain(s${CALLERID(num)})

exten => 99,1,VoiceMailMain(${CALLERID(num)},s)
exten => 1010,1,Goto(wg-menu,s,1)

exten => 3333,hint,SIP/3333
exten => 4444,hint,SIP/4444
exten => 5555,hint,SIP/5555
exten => 6666,hint,SIP/6666
exten => 7777,hint,SIP/7777
exten => 8888,hint,SIP/8888
exten => 9999,hint,SIP/9999

[extern238]
exten => _X.,1,Dial(SIP/5555&SIP/4444, 10)
exten => _X.,n,Dial(SIP/5555&SIP/4444&SIP/6666, 10)
exten => _32843287,1,Dial(SIP/9999)


[extern410]
exten => _X.,1,Dial(SIP/6666)
exten => _41006173,1,Goto(wg-menu,s,1)

[wg-menu]
exten => s,1,Wait(1)
exten => s,2,Set(TIMEOUT(digit)=5)
exten => s,3,Set(TIMEOUT(response)=15)
exten => s,4,BackGround(welcome) ; wobei ich hier einfach entweder selber aufnehmen w�rde oder ein fertieges TTP wie z.B: ;http://www.research.att.com/~ttsweb/tts/demo.php
exten => s,5,Waitexten(20)
exten => s,6,Goto(s,4)

exten => 5,1,Goto(call,5555,1)
exten => 6,1,Goto(call,6666,1)
exten => 3,1,Goto(call,3333,1)

[call]
;exten => 5555,1,Dial(SIP/5555)
;exten => 5555,2,Congestion
;exten => 5555,3,Hangup

;exten => 6666,1,Dial(SIP/6666)
;exten => 6666,2,Congestion

[mainmenu]
  exten => s,1,Answer
  exten => s,2,SetMusicOnHold(default)
  exten => s,3,Set(TIMEOUT(digit)=5)
  exten => s,4,ResponseTimeout,10
  ;SAI menu - 1 for tech support, 2 for voicemail, 3 for echo test
  exten => s,5,Background(sai-welcome)
  exten => s,6,Background(sai-choose)

 exten => 2,1,VoicemailMain()
  exten => 2,2,Hangup