Category Archives: IT/Telefoni

[featuremap]
;blindxfer => #1                ; Blind transfer  (default is #)
blindxfer => *2   ; Flytta koden till *2

exten => s,n,Dial(SIP/tmm&SIP/mobil&SIP/tmm-vood&SIP/megatmm,30,wWkKt)
exten => s,n,Goto(s-${DIALSTATUS},1)            ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => s-NOANSWER,1,VoiceMail(1001,u)
exten => s,n,Gotoif($${BLINDTRANSFER}=""?500)
exten => s,500,Hangup
exten => s,501,Set(tx=${BLINDTRANSFER:0:7})
exten => s,502,SetCallerID(RET_${EXTEN} <${CALLERIDNUM}>)
exten => s,503,Dial(${tx},,mTt)
exten => s,504,Hangup

[general]
  context=default                                                ; Default context for incoming calls
  allowoverlap=no                                                ; Disable overlap dialing support. (Default is yes)
  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
  realm=asterisk
  domain=my.sip.domain                     ; Set default domain for this host
                                                                        ; If configured, Asterisk will only allow
                                                                        ; INVITE and REFER to non-local domains
                                                                        ; Use "sip show domains" to list local domains
  ;disallow=all                                    ; First disallow all codecs
  allow=ulaw                                     ; Allow codecs in order of preference
  allow=ulaw
  allow=alaw
  allow=g729
  rtptimeout=60                                                  ; Terminate call if 60 seconds of no RTP or RTCP activity
  rtpholdtimeout=300                          ; Terminate call if 300 seconds of no RTP or RTCP activity
   
  ; Be careful with this
  sipdebug = yes                                                 ; Turn on SIP debugging by default, from
  recordhistory=yes                            ; Record SIP history by default 
  dumphistory=yes                             ; Dump SIP history at end of SIP dialogue
   
  ;; SIP Phone 1
  register => username.megaphone.se:password@sip.megaphone.se
   
  ;; SIP Phone 2
  ; /1001 = Connect immediately to extension 1001 and pass by any automated messages...
  ; (this seem to be linked to the extension-config)
  register => username.megaphone.se:password@sip.megaphone.se/1001
   
  localnet=10.0.0.0/255.0.0.0
  localnet=192.168.0.0/255.255.0.0
  nat=yes                                                           ; Global NAT settings  (Affects all peers and users)
  canreinvite=no                                ; Asterisk by default tries to redirect the
   
  [authentication]
  ; Obviously not required (Dual stuff)
  ;;;auth=username.megaphone.se:password@sip.megaphone.se
  ;;;auth=username.megaphone.se:password@sip.megaphone.se
   
  ;;;; USER CONFIGURATION
   
  ; Main extension for home
  [self]
  type=friend
  username=me
  secret=mypassword
  host=dynamic
  context=myextension
  regexten=1001
   
  ; Let the Vood have its own extension if this runs on SIP
  [self-vood]
  type=friend
  username=self-vood
  secret=voodpassword
  host=dynamic
  context=myextension
  regexten=1011
   
  ; Let the Nokia on!
  [selfmobile]
  type=friend
  username=selfmobile
  secret=mobilepassword
  host=dynamic
  context=myextension
  regexten=1010
   
  [user1]
  type=friend
  username=user1
  secret=password1
  host=dynamic
  context=user1
  regexten=2000
   
  [user2]
  type=friend
  username=user2
  secret=password2
  host=dynamic
  context=user2
  regexten=2001
   
   
  ; The PEERS (Megaphone)
  [NUM1]
  type=peer
  username=username.megaphone.se
  fromdomain=sip.megaphone.se
  insecure=very
  qualify=yes
  nat=yes
  secret=password
  host=sip.megaphone.se
  context=default
   
  [NUM2]
  type=peer
  username=username.megaphone.se
  fromdomain=sip.megaphone.se
  insecure=very
  qualify=yes
  nat=yes
  secret=password
  host=sip.megaphone.se
  context=default
  

 
  [general]
  static=yes
  writeprotect=no
  clearglobalvars=no
  priorityjumping=yes
  ;#include "filename.conf"
  [globals]
  CONSOLE=Console/dsp                                                        ; Console interface for demo
  ;TRUNK=Zap/G2                                                                                  ; Trunk interface
   
  ; Never required... (This is dual number!)
  ;;;SIPtrunk=SIP/username.megaphone.se:password@sip.megaphone.se    ; SIP trunk
  ;;;SIPtrunk=SIP/username.megaphone.se:password@sip.megaphone.se    ; SIP trunk
   
  ; Used by everybody! (Incoming connections)
  [default]
  include => allextensions
  exten => s,1,NoOp(${SIP_HEADER})
  exten => s,n,NoOp(${CALLERID})
  exten => s,n,Wait(1)
  exten => s,n,Answer
  ; Play a message!
  exten => s,n,BackGround(welcome)
  ; Wait for a beep for about 2 sec!
  exten => s,n,WaitExten(2)
  ; Play another message (in this case: "redirecting your call")
  exten => s,n,BackGround(calling_dittsamtal)
  ; If no extension set above, dial those number for 30 sec
  exten => s,n,Dial(SIP/self&SIP/selfmobile&SIP/self-vood,30)
  ; Then check status...
  exten => s,n,Goto(s-${DIALSTATUS},1)                                 ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
  ; And send to voicemail for extension 1001 on no answer
  exten => s-NOANSWER,1,VoiceMail(1001,u)
   
  [self]
  include => default
  include => outgoing
   
  [self-vood]
  include => default
  include => allextensions
  include => outgoing
   
  [selfmobile]
  include => default
  include => outgoing
   
  ; Only allow internal calls for user 1
  [user1]
  include => allextensions
  exten => 1000,1,Voicemailmain(1000)
   
  ; Allow the same as "self" for user2
  [user2]
  include => allextensions
  include => outgoing
  exten => 1000,1,Voicemailmain(1000)
  exten => _X.,1,Dial(SIP/${EXTEN}@NUM1,30,r) 
   
  ; All outgoing extensions
  [outgoing]
  ; Allow yourself to dial between your external extensions
  exten => NUM1,1,Dial(SIP/${EXTEN}@NUM2,30,r)
  exten => NUM2,1,Dial(SIP/${EXTEN}@NUM1,30,r)
  ; Anything else goes out through NUM1
  exten => _X.,1,Dial(SIP/${EXTEN}@NUM1,30,r) 
  ; But if any call starts with an extra zero, use NUM2
  ; This makes both numbers available for outbound traffic
  exten => _00X.,1,Dial(SIP/${EXTEN:1}@NUM2,30,r)
  exten => s,n,Goto(s-${DIALSTATUS},1)                                 ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
  exten => s-NOANSWER,n,VoiceMail(${ARG1},u)
   
  ; Internal usage
  [allextensions]
  ; Extension 1000 - Voicemail!
  exten => 1000,1,Voicemailmain(1000)
   
  ; Self extension and the default! Try for 20 sec
  exten => 1001,1,Dial(SIP/self&SIP/self-vood&SIP/selfmobile,20)
  ; Missed calls goes to voicemail
  exten => 1001,2,VoiceMail(1001,u)
   
  ; Others
  exten => 1010,1,Dial(SIP/selfmobile,20)
  exten => 1010,2,VoiceMail(1001,u)
  exten => 1011,1,Dial(SIP/self-vood,20)
  exten => 1011,2,VoiceMail(1001,u)
   
  ; User 1
  exten => 2000,1,Dial(SIP/user1,20)
  exten => 2000,2,VoiceMail(2000,u)
   
  ; User 2
  exten => 2001,1,Dial(SIP/user2,20)
  exten => 2001,2,VoiceMail(2001,u)
   
  ; Invalid extensions goes to voicemail
  exten => i,1,VoiceMail(1001,u)
  

To: "0435xxxxxx"

exten => s,n,Set(CALLERID(num)=${CALLERID(num)} to 0435xxxxxx)

0812345677 to 0435123456@127.0.0.1

register => username.megaphone.se:password@sip.megaphone.se/1001

register => username.megaphone.se:password@sip.megaphone.se/1020

exten => 1020,1,Set(CALLERID(num)=${CALLERID(num)} to 0435xxxxxx)
  exten => 1020,n,Dial(SIP/self&SIP/self-vood&SIP/selfmobile,20)

0812345677 to 0435654321@127.0.0.1