SPA3102 with asterisk
After asterisk 12, we use pjsip instead of sip.
Following is a pjsip.conf with template used.
;===============TRANSPORT
[simpletrans]
type=transport
protocol=udp
bind=0.0.0.0
;===============ENDPOINT TEMPLATES
[endpoint-basic](!)
type=endpoint
transport=simpletrans
context=internal
disallow=all
allow=ulaw,alaw
[auth-userpass](!)
type=auth
auth_type=userpass
[aor-single-reg](!)
type=aor
max_contacts=1
;===============EXTENSION fxs
[200](endpoint-basic)
auth=auth200
aors=200
[auth200](auth-userpass)
password=123456
username=200
[200](aor-single-reg)
;===============EXTENSION fxo
[fxo](endpoint-basic)
auth=authfxo
aors=fxo
[authfxo](auth-userpass)
password=123456
username=fxo
[fxo](aor-single-reg)
;=================== EXTENSION ip
[201](endpoint-basic)
auth=auth201
aors=201
[auth201](auth-userpass)
password=123
username=201
[201](aor-single-reg)
Following is a extension.conf(Dial Plan):
[internal]
exten => 200,1,Dial(PJSIP/200)
exten => 201,1,Dial(PJSIP/201)
exten => _0.,1,Dial(PJSIP/${EXTEN:1}@fxo)
;phone number that start with 0 are sent to Linksys -> landline
;exten => group,1,Dial(PJSIP/200,PJSIP/201)
exten => group,1,Dial(PJSIP/200, 30, m)
same => n,Answer()
same => n,Wait(1)
same => n,Background(custom-menu)
same => n,WaitExten(10)
same => n,Hangup()
exten => 1,1,VoiceMail(1234@default)
same => n,Hangup()
exten => 5555,1,Answer(500)
same => n,Record(en/custom-menu.gsm)
same => n,Wait(1)
same => n,Playback(custom-menu)
same => n,Hangup()
exten => 9999,1,VoiceMailMain(1234@default)
same => n,Hangup()
The same conf which use chan_sip is like following:
[fxo]
type=friend
secret=123
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=internal
[200]
type=friend
secret=123
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=internal
[201]
type=friend
secret=123
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=internal
For SPA3102, we should notice the Dial Plans of PSTN line: (S0<:group@my.asterisk.server>) (Dial group on my asterisk server)