Line Indikation LED

exten => _**.,1,PickupChan(SIP/${EXTEN:2})

[B]Multi Purpose Key 1:[/B] Key Mode: [Asterisk BLF]     Account: [Account 1]
                         Name: [Mustermann     ]  UserID [884     ]

exten => 884,hint,SIP/884

*CLI> sip show settings


Global Settings:
----------------
  SIP Port:               5060
  Bindaddress:            0.0.0.0
  Videosupport:           No
  AutoCreatePeer:         No
  Allow unknown access:   Yes
  Promsic. redir:         No
  SIP domain support:     No
  Call to non-local dom.: Yes
  URI user is phone no:   No
  Our auth realm          asterisk
  Realm. auth:            No
  User Agent:             Asterisk PBX
  MWI checking interval:  10 secs
  Reg. context:           (not set)
  Caller ID:              asterisk
  From: Domain:
  Record SIP history:     Off
  Call Events:            Off
  IP ToS:                 0x0
  OSP Support:            No
  SIP realtime:           Disabled

Global Signalling Settings:
---------------------------
  Codecs:                 alaw,ulaw,gsm,ilbc
  Relax DTMF:             No
  Compact SIP headers:    No
  RTP Timeout:            0 (Disabled)
  RTP Hold Timeout:       0 (Disabled)
  MWI NOTIFY mime type:   application/simple-message-summary
  DNS SRV lookup:         Yes
  Pedantic SIP support:   No
  Reg. max duration:      3600 secs
  Reg. default duration:  120 secs
  Outbound reg. timeout:  20 secs
  Outbound reg. attempts: 0
  Notify ringing state:   Yes

Default Settings:
-----------------
  Context:                default
  Nat:                    RFC3581
  DTMF:                   rfc2833
  Qualify:                0
  Use ClientCode:         No
  Progress inband:        Never
  Language:               de
  Musicclass:             default
  Voice Mail Extension:   asterisk

[general]
context=default		; Default context for incoming calls
bindport=5060		; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0		; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes		; Enable DNS SRV lookups on outbound calls, Note: Asterisk only uses the first host in SRV records Disabling DNS SRV lookups disables the ability to place SIP calls based on domain names to some other SIP users on the Internet

maxexpiry=3600		; Max length of incoming registration we allow
defaultexpiry=120		; Default length of incoming/outoing registration
;notifymimetype=text/plain	; Allow overriding of mime type in MWI NOTIFY
;checkmwi=10			; Default time between mailbox checks for peers
;vmexten=voicemail      ; dialplan extension to reach mailbox sets the Message-Account in the MWI notify message defaults to "asterisk"

disallow=all			; First disallow all codecs
allow=alaw
allow=ulaw				; Allow codecs in order of preference
allow=gsm
allow=ilbc 
;allow = g729
language=de				; Default language setting for all users/peers, this may also be set for individual users/peers
subscribecontext=default	; Set a specific context for SUBSCRIBE requests
					; Useful to limit subscriptions to local extensions
					; Settable per peer/user also
;notifyringing=yes			; Notify subscriptions on RINGING state

[34]
callerid=Martin 34 <34>		; Full caller ID, to override the phones config
host=dynamic
domain=mx.beauty-case.de
nat=yes
nat=no				; there is not NAT between phone and Asterisk
type=peer
;type=friend
user=34
secret=xxx
mailbox=34@default		; mailbox in voicemail context "default"
canreinvite=no
;canreinvite=yes			; allow RTP voice traffic to bypass Asterisk
callgroup=1
pickupgroup=1 
incominglimit=1
;call-limit=1			; permit only 1 outgoing call and 1 incoming call at a time
					; from the phone to asterisk
					; (1 for the explicit peer, 1 for the explicit user,
					; remember that a friend equals 1 peer and 1 user in
					; memory)
context=sip-lokal			; testen ob ohne möglich ,da global
qualify=1000
subscribecontext=default	; testen ob ohne möglich, da global
regexten=34

[33]
callerid=Martin 2 <33>
host=dynamic
domain=mx.beauty-case.de
nat=yes
;type=peer
type=friend
user=33
secret=xxx
mailbox=33
canreinvite=yes
callgroup=1
pickupgroup=1
;incominglimit=1
call-limit=2
context=sip-lokal
qualify=1000
subscribecontext=default
;regexten=33

[31]
callerid=Martin <31>
host=dynamic
domain=mx.beauty-case.de
nat=yes
;type=peer
type=friend
user=31
secret=yes
mailbox=31
canreinvite=yes
callgroup=1
pickupgroup=1
;incominglimit=1
call-limit=2
context=sip-lokal
qualify=yes
subscribecontext=default
regexten=31
;regexten=666		; When they register, create extension 666

asterix*CLI> sip show settings

Global Settings:
----------------
  SIP Port:               5060
  Bindaddress:            0.0.0.0
  Videosupport:           No
  AutoCreatePeer:         No
  Allow unknown access:   Yes
  Promsic. redir:         No
  SIP domain support:     No
  Call to non-local dom.: Yes
  URI user is phone no:   No
  Our auth realm          sip.in.rudolph-log.de
  Realm. auth:            No
  User Agent:             Asterisk PBX
  MWI checking interval:  10 secs
  Reg. context:           (not set)
  Caller ID:              asterisk
  From: Domain:
  Record SIP history:     Off
  Call Events:            Off
  IP ToS:                 0x0
  OSP Support:            No
  SIP realtime:           Disabled

Global Signalling Settings:
---------------------------
  Codecs:                 ulaw
  Relax DTMF:             No
  Compact SIP headers:    No
  RTP Timeout:            0 (Disabled)
  RTP Hold Timeout:       0 (Disabled)
  MWI NOTIFY mime type:   application/simple-message-summary
  DNS SRV lookup:         Yes
  Pedantic SIP support:   No
  Reg. max duration:      3600 secs
  Reg. default duration:  120 secs
  Outbound reg. timeout:  20 secs
  Outbound reg. attempts: 0
  Notify ringing state:   Yes

Default Settings:
-----------------
  Context:                default
  Nat:                    RFC3581
  DTMF:                   inband
  Qualify:                0
  Use ClientCode:         No
  Progress inband:        Never
  Language:               de
  Musicclass:             default
  Voice Mail Extension:   asterisk

----

[general]
context=default
realm=sip.in.rudolph-log.de
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
canreinvite=no
language=de
dtmfmode=inband
disallow=all
allow=ulaw
canreinvite=no

[884]
type=friend
host=dynamic
secret=884
callgroup=1
pickupgroup=1
mailbox=884@default

[888]
type=friend
host=dynamic
secret=888
callgroup=1
pickupgroup=1
mailbox=888@default

[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no

[globals]
CONSOLE=Console/dsp             ; Console interface for demo
TRUNK=Zap/g1                    ; Trunk interface
TRUNKMSD=1                      ; MSD digits to strip (usually 1 or 0)
ignorepat => 0

[default]
exten => 8500,1,VoicemailMain()
exten => 8500,n,Hangup()
exten => 8500,n,VoicemailMain(${CALLERIDNUM})
exten => 8500,n,Hangup()

exten => _ZXX,1,NoOp(interners Gespraech ${CALLERIDNUM}->${EXTEN})
exten => _ZXX,n,GotoIf($[ ${CALLERIDNUM} = ${EXTEN} ]?8500,3) ; VM?
exten => _ZXX,n,Dial(SIP/${EXTEN},30,tT)
exten => _ZXX,n,Voicemail(u${EXTEN}@default)
exten => _ZXX,104,Voicemail(b${EXTEN}@default)

exten => 888,hint,SIP/888
exten => 884,hint,SIP/884
exten => 885,hint,SIP/885

exten => _**ZXX,1,NoOp(Pickup von ${EXTEN:2} durch ${CALLERIDNUM})
exten => _**ZXX,n,PickupChan(SIP/${EXTEN:2})