[asterisk-users] Voicemail/dtmf not working?

[Alan Lord]

Hi,

I am setting up an asterisk server for testing purposes and cannot get 
voicemail to work at all.

My host OS is Linux From Scratch 6.3 and the asterisk software versions 
I built are zaptel-1.4.5.1 and asterisk-1.4.12.

I am using the Ekiga softphone on my Ubuntu desktop machine. My asterisk 
server and client phone are on different computers but are on the same 
LAN, i.e. no NAT.

I have an "echo test" extension which works fine. But when I try to call 
into my voicemail account using 8100, I do not hear the first "Playing 
'vm_password'" message (although do I hear the subsequent messages). And 
any numbers I enter on the ekiga keypad do not seem to be recognised by 
asterisk (I enabled debug rdp and rfc2833 packets appear to be 
transmitted and received). The softphone is set up to use RFC2833 for DTMF.

I have a single x100p card installed which seems to work - to a 
fashion... Incoming calls are answered and the greeting is heard, but 
the line hangs up instantly the message finishes. (A different problem 
which I will investigate seperately unless someone has a quick answer). 
Outgoing calls seem to be O.K. An lsmod of my system reveals the following:
===================================================
Module                  Size  Used by
zttranscode             6280  0
wcfxo                   9760  0
zaptel                186660  6 zttranscode,wcfxo
crc_ccitt               1792  1 zaptel
===================================================

Below is a typical call log (I *am* typing 1234 on the ekiga keypad 
during this call) and my extension, sip and voicemail.conf files. Anyone 
got any suggestions?

Messages on log
=========================================
     -- Executing [8100 at internal:1] Answer("SIP/100-081d9478", "") in 
new stack
     -- Executing [8100 at internal:1] VoiceMailMain("SIP/100-081d9478", 
"100") in new stack
     -- <SIP/100-081d9478> Playing 'vm-password' (language 'en')
     -- Incorrect password '' for user '100' (context = default)
     -- <SIP/100-081d9478> Playing 'vm-incorrect' (language 'en')
     -- <SIP/100-081d9478> Playing 'vm-password' (language 'en')
     -- Incorrect password '' for user '100' (context = default)
     -- <SIP/100-081d9478> Playing 'vm-incorrect' (language 'en')
     -- <SIP/100-081d9478> Playing 'vm-password' (language 'en')
     -- Incorrect password '' for user '100' (context = default)
     -- <SIP/100-081d9478> Playing 'vm-incorrect' (language 'en')
     -- <SIP/100-081d9478> Playing 'vm-goodbye' (language 'en')
   == Auto fallthrough, channel 'SIP/100-081d9478' status is 'UNKNOWN'
=========================================


extension.conf
========================================
;exten => $name,$priority,$application()
[globals]
ALANL=SIP/100
OUTBOUNDTRUNK=Zap/1
FWDNUMBER=867*** ; My FreeWorldDialup Number
FWDCIDNAME="Alan Lord" ; My CLI
FWDPASSWORD=******
FWDRINGS=${ALANL} ; Phone to ring
FWDVMBOX=1000 ; Voice Mail Box (not yet setup)

[zap_incoming] ; Channel defined in zapata.conf
exten => s,1,Answer( )
exten => s,2,Set(TIMEOUT(digit)=5)
exten => s,3,Set(TIMEOUT(response)=30)
exten => s,4,Background(vm-enter-num-to-call)
exten => s.5,Wait(5) ;Try to stop line hanging up straight away - failed

exten => t,1,Goto(s,2) ; Repeat s,2 if no input from caller

exten => i,1,Playback(pbx-invalid)
exten => i,2,Goto(s,2)

exten => 100,1,Dial(${ALANL},10)
exten => 100,2,VoiceMail(u100)
exten => 100,102,VoiceMail(b100)
exten => 100,3,Hangup()

[internal]
include => outbound-local

; My ekiga SoftPhone
exten => 100,1,Dial(${ALANL},,r)

;Outbound to FreeWorlDialup
exten => _393.,1,SetCallerId,${FWDCIDNAME}
exten => 
_393.,2,Dial(IAX2/${FWDNUMBER}:${FWDPASSWORD}@iax2.fwdnet.net/${EXTEN:3},60,r)
exten => _393.,3,Congestion

; Local echo test
exten => 611,1,Answer()
exten => 611,2,PlayBack(demo-echotest)
exten => 611,3,Echo()
exten => 611,4,PlayBack(demo-echodone)
exten => 611,5,Hangup()

; Manage Voicemail
exten => _8XXX,1,Answer()
exten => _8XXX,2,VoiceMailMain(${EXTEN:1})

; Outbound via PSTN
[outbound-local]
exten => _9XXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten => _9XXXXXX,2,Congestion()
exten => _9XXXXXX,102,Congestion()

exten => 999,1,Dial(${OUTBOUNDTRUNK}/999)
exten => 9999,1,Dial(${OUTBOUNDTRUNK}/999)

[fromiax] ; IAX trunk from Alan B defined in iax.conf
;TBD

[fromiaxfwd] ;IAX Trunk from FWD
exten => ${FWDNUMBER},1,Dial(${FWDRINGS},20,r)
exten => ${FWDNUMBER},2,Voicemail,u${FWDVMBOX}
exten => ${FWDNUMBER},102,Voicemail,b${FWDVMBOX}
=================================================

sip.conf
=================================================
[general]
srvlookup=yes
disallow=all
allow=alaw
allow=ulaw
allow=gsm
dtmfmode=auto

[100]
type=friend
callerid=Alan Lord
secret=******
qualify=yes ; Qualify peer is no more than 2000 ms away
nat=no ; This phone is not natted
host=dynamic ; This device registers with us
canreinvite=no ; Asterisk by default tries to redirect
context=internal ; the internal context controls what we can do
mailbox=100 ; Voicemail Box
===================================================

voicemail.conf
===================================================
[general]
; Send voice file attachments in email notifications
attach=yes

; Use wav49 format for all voicemail messages
format=wav49

; Limit the maximum message length to 180 seconds
maxmessage=180

; Limit the minimum message length to 3 seconds
minmessage=3

; Announce caller's number before playing message
;saycid=yes

; Limit the login retiries to 3 before disconnecting the caller
maxlogins=3

[default]
100=>1234,Alan Lord,,,
===================================================


Anyone got any ideas?

Cheers

Alan

-- 
The way out is open!
http://www.theopensourcerer.com