I do indeed have a Toshiba 424 like yours and have it integrated with Asterisk. I have a couple different integrations.


On the asterisk box I have the following.

1 - TE410 card with quad T1's.

Interface 1 connected to the Toshiba with E&M to a RDTU card on the Toshiba Interface 2 connected to 24 ports ADIT600 fxo channels. These channels are then tied into the toshiba analog extensions Interface 3 connected to 24 ports of ADIT600 fxs channels. This is to provide FXS (modem, fax, dialup, etc) throughout our building.
Interface 4 connected to one of our 4 dedicated long distance T1's with Qwest. (The other 3 are connected through the Toshiba)

The biggest challenge that I have had with the Toshiba -> asterisk integration is the Toshiba's inability to do unsupervised trunk to trunk transfers. That means if the call originates from the Toshiba, you are unable to "transfer" it over to the Asterisk box via an E&M T1 circuit. The other way around is no problem.

In your case, if you are adding voicemail, you will probable want to look at an Adit 600 like I use. It would interface to analog extensions on your Toshiba and the Adit would convert it to Digital on the Asterisk via a Digium card. TE100 should work for this.

One of the toughest things for you will be the disconnect supervision.
The Toshiba doesn't send a standard disconnect. It has a proprietary DTMF disconnect that it sends the Strata voicemail systems, but I'm not sure what tone it is. It's on my to-do list to figure out.
Otherwise you are going to have to use busy supervision in Asterisk which will leave several seconds of Congestion on the back of every voicemail.

This may help for the above issue... Allow the dtmf 'D' tone sent by a Toshiba DK424. Turn on LEDs 15, 19 & 20 in program-31 for the toshiba's voicemail ports.
By passing the audio through Asterisk's MeetMe application, it can be made to listen for any dtmf tone. This setup defined here just hangs-up when a 'A', 'B' or 'D' tone arrives but minor modifications (see the 'exten => _[ABD#],1,Hangup()' line) opens up the possibility of more full featured voicemail integration.
Now a couple of notes:
1) Voicmail ports arriving into asterisk are all two digits. You may have to make modifications to use this. They arrive in the vm_from_toshiba context (as defined in zapata.conf).
2) All of my Toshiba extension are in the form 7XXX
3) Regard this as a rough guide - but these snips work in my configuration.

;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;

In Zapata.conf

callwaitingcallerid=no
threewaycalling=no
transfer=no
usecallerid=no
signalling=fxs_ks
callerid="Voicemail HS trunks" <7000>
context=vm_from_toshiba
group=10
channel => 49-56 ; These are the two digit ports mentioned above. The digit number is important where the ${CHANNEL} variable is referenced.

;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;

These contexts are defined in extensions.conf

[vm_from_toshiba]
exten => s,1,NoCDR()
exten => s,n,wait(.5)
exten => s,n,answer
exten => s,n,SET(TIMEOUT(absolute)=1800)
exten => s,n,SET(TIMEOUT(digit)=1.5)
exten => s,n,SET(TIMEOUT(response)=7)
exten => s,n,waitexten(3)
exten => s,n,VoicemailMain()
exten => s,n,hangup
exten => _X.,1,noop(Dialed extension is ${EXTEN})
exten => _X.,n,Gotoif($["${EXTEN:-1}" = "D"]?terminated) ;;Checkout Toshiba seems to be sending a terminator 'D' There may have been a hangup before the whole dtmf string was transmitted.
exten => _X.,n,System(/root/toshiba_error_finder ${EXTEN} &) ;;; This calls a little debugging script - comment it out
exten => _X.,n,Voicemail(9999,u) ;;; There is a mistake on the toshiba forwarding string, let the caller tell me who they were trying to call - comment it out.
exten => _X.,n,wait(1.5)
exten => _X.,n(hu),hangup
exten => _X.,n(terminated),NoCDR()
exten => _X.,n,Voicemail(9999,u)
exten => _X.,n,hangup
exten => _917XXX,1,SET(TIMEOUT(absolute)=300) ;;; max msg 5 min - adjust as needed
exten => _917XXX,n,Goto(hs-vm-with-dtmf-detect,${EXTEN:2},1)
exten => _917XXX,n,Hangup()
exten => _91XXXX,1,System(touch /tmp/vm/old_ext/${EXTEN:2}) ;;;; This is for debuging too
exten => _91XXXX,n,Hangup()
exten => _927XXX,1,VoicemailMain(${EXTEN:2}|s))
exten => _927XXX,n,Hangup()
exten => _92XXXX,1,System(touch /tmp/vm/92/${EXTEN})
exten => _92XXXX,n,Hangup()
exten => T,1,Hangup()

[hs-vm-with-dtmf-detect]
exten => _7XXX,1,System(asterisk -rx "originate Local/${CHANNEL:4:2}@hs-vm-meetme application voicemail \\"${EXTEN}|su\\"")
exten => _7XXX,n,MeetMe(vmd${CHANNEL:4:2},AdFpqXx) ;Meetme mail boxes defined for zap channels 49-56
exten => _7XXX,n,Hangup()
exten => _[ABD#],1,Hangup()
exten => h,1,MeetMeAdmin(vmd${CHANNEL:4:2},K) ;Kick all users out of conference

[hs-vm-meetme]
exten => _X.,1,NoCDR()
exten => _X.,n,Wait(.3) ; give hs-vm-with-dtmf-detect time to create meetme
exten => _X.,n,MeetMe(vmd${EXTEN},q)
exten => _X.,n,Hangup()

[vm_toshiba_log_mwi]
exten => s,1,wait(1.5)
exten => s,n,hangup

[vm_toshiba_set_mwi]
exten => _7XXX,1,Gotoif(${MWI_ON}?on)
exten => _7XXX,n(off),Dial(Zap/52/!#64${EXTEN}|30)
exten => _7XXX,n,wait(1)
exten => _7XXX,n,hangup()
exten => _7XXX,n(on),Dial(Zap/52/!#63${EXTEN}|30)
exten => _7XXX,n,wait(1)
exten => _7XXX,n,hangup()
exten => _x.,1,wait(1)
exten => _x.,n,hangup()

;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;

In Voicemail.conf:
externnotify = /usr/local/bin/toshiba_notify

;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;

Finally here is the bash script referenced in externalnotify, toshiba_notify (READ THE COMMENTS!):
#!/bin/bash
# Written by John Harragin, Monroe-Woodbury Central-School-District 845-460-6200 x6265
CONTEXT=$1
VM_EXTEN=$2
NUM_MSG=$3
TMP_CALL_FILE=/tmp/vm/${VM_EXTEN}.call
REAL_CALL_FILE=/var/spool/asterisk/outgoing/${VM_EXTEN}.call
# I only want to send notification to phones hosted on the toshiba - not SIP phones.
# if you use this script, I suggest geting rid of the next 4 lines and the fi on the last line - or modify to you needs.
if [ $(grep -E "\[${VM_EXTEN}\]" /etc/asterisk/sip_phones/ConsolidatedSip-70.conf) ] && [ ${CONTEXT} != "default" ];then
echo $(grep -E "\[${VM_EXTEN}\]" /etc/asterisk/sip_phones/ConsolidatedSip-70.conf)
 echo "$(date) context=${CONTEXT} VM_EXTEN=${VM_EXTEN}  NUM_MSG=${NUM_MSG} - not a HS ext, ignoring." >> /var/log/asterisk/toshiba_vm.log
else
if [ ${NUM_MSG} -le 1 ];then
echo "Channel: Local/${VM_EXTEN}@vm_toshiba_set_mwi
MaxRetries: 4
RetryTime: 90
WaitTime: 30
SET: VM_EXTEN=${VM_EXTEN}
SET: MWI_ON=${NUM_MSG}
Context: vm_toshiba_log_mwi
Extension: s
Priority: 1" > ${TMP_CALL_FILE}
mv ${TMP_CALL_FILE} ${REAL_CALL_FILE}
fi
echo "$(date) context=${CONTEXT} VM_EXTEN=${VM_EXTEN} NUM_MSG=${NUM_MSG}" >> /var/log/asterisk/toshiba_vm.log
fi # This is the line to delete if you read the upper comment.

I currently have asterisk connected to the a 4 port T1 card. The Voice T1 line goes into asterisk, then a T1 cross connect cable connected to asterisk and the Toshiba CTX. I only have 3 phones, and 7 analog lines connected to the Toshiba, the rest are Polycom IP 500 phones connected to asterisk.

Yes, you are correct. I have the T1 line from the Strata plugged into the T1 card on the asterisk system with a T1 crossover cable. Then in Zapata.conf I added the following lines for this connection.

context=from-ctx
switchtype=national
pridialplan=unknown
signalling=pri_net
echocancel=yes
overlapdial=no
immediate=no
echocancelwhenbridged=yes
echotraining=yes
rxgain=-3.0
group=1
callprogress=no
channel => 25-47

Channels 1-24 are my phone company T1, 25-47 go to the Strata. In my Zaptel.conf I have

span=2,2,0,esf,b8zs
bchan=25-47
dchan=48

If you make a correct crossover cable, the light should be green on your asterisk T1 card. Then to call from Asterisk to the Strata I use the following in my extensions.conf (assuming number 6050)

Exten => 6050,1,Dial(ZAP/g1/${EXTEN})

This will move the call to the Strata. If the strata was the only t1 line, you would use channel 1-23 and ZAP/g0 and span=1,1,0,esf,b8zs with 24 as the dchan.

When 6050 is dialed, the strata thinks it is an incoming call from the phone company. You would of course need an extension 6050 in CTX winadmin, and the strata would ring the phone associated with it.

The Strata thinks it is talking to the phone company, the Asterisk box is completely transparent. As to calls from the Strata, you would need to match those in your asterisk dial plan.

I.E. If you dial 96018 from the strata, you would need an Exten => 6018,1,Dial(SIP/6018,15)
In this example you would be calling a SIP phone connected to asterisk.

The 9 assumes you need to dial a 9 to get out of the Strata.

Now going back to my system having the Real phone T1 connected at ZAP/g0 , I used ctx-out as the context, if I wanted to call someone outside the company from the strata I would need something like this in my extensions.conf

ctx-out

;check for internal extension

Include => from-internal

;call outside

Exten => XXXXXX.,1,Dial(ZAP/g0/${EXTEN})
Exten => XXXXXX.,2,Congestion

This would place the call out, and phones connected to the Strata would call out as though it was connected to the phone company.

exten => _X.,1,Set(CALLERID(name)=)