;SpoonFeed 3000   (greg@nufone.net)  http://www.nufone.net

;

; Example extensions.conf
[local_phone]
exten => _1NXXNXXXXXX,1,Dial,Zap/g1/${EXTEN}||Ttr ; make sure u have the channel set in zapata.conf
exten => _NXXXXXX,1,Dial,SIP/pstn/${EXTEN}||Ttr  ; spa3k

[incoming]
exten => s,1,Answer
exten => s,2,Wait(1)
exten => s,3,Set(TIMEOUT(digit)=3)
exten => s,4,Set(TIMEOUT(response)=5)
exten => s,5,BackGround(kimosabe-greeting) ;greeting  shidoshome_greet)

exten => t,1,Goto(kimosabe_phones|1000|1) ; or wherever you want to send it

exten => i,1,Playback,pbx-invalid
exten => i,2,Goto(kimosabe_phones|1000|1) ; or wherever you want to send it


include => kimosabe_phones
include => voicemail

[kimosabe_phones]
exten => _1000,1,Playback,transfer
exten => _1000,2,Dial(SIP/1000|15|Tmp)
exten => _1000,3,Voicemail,u1000
exten => _1000,103,Voicemail,b1000

[voicemail]
exten => _69XXXX,1,VoicemailMain,${EXTEN:2}
exten => _69XXXX,2,Hangup
exten => 6969,1,VoicemailMain


; Example sip.conf


[kimosabe1]
mailbox=1000                    ; Mailbox number (numerical so can be selected through phone)
type=friend
host=dynamic
secret=foo
call-limit=2
disallow=all                    ; need to disallow=all before we can use allow=
allow=ulaw                      ; Note: In user sections the order of codecs
context=local_phone             ; This links sip.conf with the 
				; entry local_phone in extensions.conf

[pstn]
type=friend
host=dynamic
secret=foo
call-limit=2
disallow=all                    ; need to disallow=all before we can use allow=
allow=ulaw                      ; Note: In user sections the order of codecs
                                ; listed with allow= does NOT matter!
context=incoming


Setup the SPA-3000 for outgoing VoIP calls

There are lots of configuration settings for the SPA-3000, but I think the following are the major ones which need to be set. Using the advanced web menu configuration:

In the Line 1 tab:


Line Enable: Yes
Proxy: <IP Address of Asterisk Server>
Register: Yes
User ID: spa3000
Password: foo
Dial Plan: (9#x.|91747xxxxxxx|000S0<:@gw0>|106S0<:@gw0>|1[38]xxxxxxxxS0<:@gw0>|13[1-9]xxxS0<:@gw0>|1[9]xxxxxxxxS0<:@gw0>|0[23478]xxxxxxxxS0|[2-9]xxxxxxxS0|00

n the PSTN Line tab:


Line Enable: Yes
Proxy: <IP Address of Asterisk Server>
Register: Yes
User ID: pstn
Password: foo
Dial Plan 2: (S0<:192.168.1.1>)
PSTN-To-VoIP Gateway Enable: Yes
PSTN Ring Thru Line 1: Yes
PSTN CID For VoIP CID: Yes
PSTN Caller Default DP: 2

; Example voicemail.conf


1000 => 1234,Ding Dong,some@email.address.net


; Router Config

; Forward port 5060 UDP to the IP of your Asterisk Box
; Forward port 10001 - 20000 UDP to the IP of your Asterisk Box