dear community,
i have running asterisk 1.8 on debian 8.
my setup is the following: raving.at/diagram.jpg
My sip.conf:
[general]
alwaysauthreject=yes
port=5060
bindaddr=88.198.X.X
context=sonstige
allowguest=no
externhost=dynamicdns (from Telecom Internet)
register => 4331******1: ***@***/4331******1
[4331******1]
type=friend
context=meine-telefone
secret=***
host=dynamic
[4331******2]
type=friend
context=meine-telefone
secret=***
host=dynamic
[ext-sip-account]
type=peer
context=von-voip-provider
username=***
fromuser=***
secret=***
host=***
fromdomain=***
qualify=yes
insecure=port,invite
nat=no
allowguest=no
alwaysauthreject=yes
my extensions.conf:
[sonstige]
[meine-telefone]
exten => 1,1,Dial(SIP/4331******1)
exten => 2,1,Dial(SIP/4331******2)
exten => _0[1-9].,1,SIPAddHeader(P-Preferred-Identity: sip:${CALLERID(num)}@asterisk\;user=phone)
exten => _0[1-9].,n,Dial(SIP/${EXTEN}@ext-sip-account)
[von-voip-provider]
; eingehende Verbindungen
exten => _4331******1,1,Dial(SIP/4331******1)
exten => _4331******2,1,Dial(SIP/4331******2)
; working fine for test only (playback on call)
; exten => _4331******1,1,Answer()
; exten => _4331******1,2,Playback(testsound)
; exten => _4331******1,3,Hangup
my Problem is:
if i activate the lines with the playback, the call working fine (fast).
but now i must Setup my asterisk so, that if i call the 4331******1 that the cisco phone Adapter gets the call. currently only hear “nothing”…
the cisco phone Adapter are connected to the asterisk Server. i also have tried with a stun Server but no Chance.
can anybody help me how i can Setup my asterisk-server?
thank you