;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
include => local_phone
[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
[1000]
mailbox=1000
type=friend
host=dynamic
secret=oof
disallow=all
allow=gsm
context=kimosabe_phones
[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
In 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