; Asterisk example logs for voicemail / SIP
; Kwynn Buess, kwynn.com
; in production roughly 2021/07/31
; This example posting date:
; 2021/08/22 5:30pm EDT

; I changed the following for security:
; * The phone number, in this case as assigned by the Amazon (Web Services) Chime "Voice Connector"
; * The external IP address
; * The voicemail PIN


; /etc/asterisk/extensions.conf
[from-external]
exten = +14045551234,1,Answer(200)
same  =  n,Playback(kwprompt3)
same  =  n,VoiceMail(1@vm-try1, s)

; logger.conf
[logfiles]
console = verbose(7),notice,warning,error,debug(7)
full    = verbose(7),notice,warning,error,debug(7)

; pjsip.conf
[global]
debug=yes

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0
external_media_address=123.123.123.123
external_signaling_address=123.123.123.123

[+14045551234]
type=identify
endpoint=+14045551234
match=0.0.0.0/0


[+14045551234]
type=endpoint
context=from-external
transport=transport-udp
disallow=all
allow=ulaw
aors=+14045551234


[+14045551234]
type=aor
max_contacts=10

; sip.conf
; completely commented out, but the file needs to exist until I eliminate SIP from modules.conf or some such

; voicemail.conf
[general]
format = wav49|gsm|wav
maxmsg = 9999
maxsilence = 10000

[vm-try1]
1 = 1234,Test One