[Asterisk-Users] sip.conf extensions.conf

[Mauro Locatelli]

Hi, my sip.conf and my extensions.conf :)
I hope it's useful

**SIP.CONF**

[general]
port = 5060                 ; port to bind for sip connections
bindaddr = 0.0.0.0          ; ip to bind for sip connections
context = default           ; default context for incoming sip calls
externip = 222.99.99.22     ; Your external ip
localnet = 192.168.1.0/255.255.255.0 ;localnet and mask


disallow = all              ; disallow all codecs, we want to enable,
allow=g726
allow=ulaw
allow=alaw
allow= gsm              ; what we deem is necessary
allow= ilbc
allow= speex
                            
register => 
sipphonenumber:sipphonepwd at proxy01.sipphone.com/marlow-sip  ;information 
about sipphone

[proxy01.sipphone.com]
type=friend
username=sipphonenumber
secret=sipphonepwd
host=proxy01.sipphone.com
context=sipphone
nat=1


[marlow]
callerid=("marlow" <3986>)
username=marlow
type=friend
secret=marlowpwd
host=dynamic
context=internal
canreinvite=no
nat=1

[brandon]
callerid=("brandon" <3986>)
username=brandon
type=friend
secret=brandonpwd
host=dynamic
context=internal
canreinvite=no

[david]
callerid=("david" <3988>)
username=david
type=friend
secret=davidpwd
host=dynamic
context=internal
canreinvite=no
-----------------------------------------------------------------------------------------------------------

**EXTENSIONS.CONF**

[general]
static=yes
writeprotect=no

[globals]
MARLOW_CID=brandon
MARLOW_SIPPHONE=sipphonenumber
PHONE1=SIP/marlow  ;unuseful for now it's only a try
PHONE2=SIP/brandon ;unuseful for now it's only a try
PHONE3=SIP/david   ;unuseful for now it's only a try

[internal]
  include => from-sip
  include => sipphone
  include => tollfree
  include => 3986
  include => 3987
  include => 3988
  include => voicesystem

[voicesystem]

  exten => 9999,1,VoiceMailMain(${CALLERIDNUM}) ; extension 9999 is the VM 
system,go directly to callers VM
  exten => 9999,2,Hangup


[3986]
  exten => 3986,1,Dial(SIP/marlow,20)           ; call SIP extension "marlow" 
for 60 seconds,if extension 3986 is called
  exten => 3986,2,Voicemail(u3986)              ; if we can't connect to 
"marlow" or after seconds go to the unavail VM
  exten => 3986,102,Voicemail(b3986)            ; if busy, go to the busy VM

[3987]
  exten => 3987,1,Dial(SIP/brandon,60)           ; call SIP extension 
"brandon" for 60 seconds,if extension 3986 is called
  exten => 3987,2,Voicemail(u3986)              ; if we can't connect to 
"brandon" or after seconds go to the unavail VM
  exten => 3987,102,Voicemail(b3986)            ; if busy, go to the busy VM

[3988]
  exten => 3988,1,Dial(SIP/brandon,60)           ; call SIP extension "david" 
for 60 seconds,if extension 3986 is called
  exten => 3988,2,Voicemail(u3986)              ; if we can't connect to 
"david" or after seconds go to the unavail VM
  exten => 3988,102,Voicemail(b3986)            ; if busy, go to the busy VM


[from-sip]
  ;
  ; default extension for calls from SIP
  ;
  ; calls from sipphone
  
  ;for receive call from sipphone and send it to local phone 3986 but don't 
work:( and I don't know why
   exten => marlow-sip,1,SetCIDName(SIP - ${CALLERIDNAME}) ; indicate that the 
call came through sipphone
   exten => marlow-sip,2,Dial(Local/3986 at internal/n)
  

[outbound-internal]
  ;
  ; include local extensions
  ;
  include => internal

  ;
  ; include SIP accounts
  ;
  include => sipphone

  ;
  ; include tollfree calls
  ;
  ;include => tollfree

[default]
  ; include from-sip for default. We don't use it, but it might be a good idea
  include => from-sip
  include => sipphone
  include => internal

[sipphone]
;  Official Sipphone example don't work very well
;  exten => _1747.,1,Dial(SIP/${EXTEN}@proxy01.sipphone.com)   ; set my 
callerid and name 
;  exten => _1747.,2,Playback(notavail)                            ; this did 
not work out
;  exten => _1747.,3,Busy
  
;Approach to gateway guide
  exten => _1747.,1,SetCallerID(${MARLOW_CID} <${MARLOW_SIPPHONE}>)   ; set my 
callerid and name 
  exten => 
_1747.,2,Dial,SIP/${EXTEN:4}@proxy01.sipphone.com                      ; dial 
the number i wish to dial
  exten => _1747.,3,Playback(invalid)                            ; this did 
not work out
  exten => _1747.,4,Hangup
  exten => _1747.,103,Busy

[tollfree]
  ;
  ; terminate toll-free no.'s via fwdnet
  ;
 ;Use for call italian toll free
 ; +39 800
 ; exten => _39800.,1,SetCallerID(<${MARLOW_SIPPHONE}>)
 ; exten => _39800.,1,Dial,SIP/*${EXTEN}@proxy01.sipphone.com
 ;  exten => _39800N.,1,Dial,Zap/1/${EXTEN:2}
 
 ; Use for call external PSTN number
  exten => _0X.,1,Dial,Zap/1/${EXTEN:1}
  exten => _0X.,2,Playback(invalid)
  exten => _0X.,3,Hangup
  exten => _0X.,103,Busy
 
 ;Use for call american Toll free
 ; +1-800
  exten => _1800.,1,SetCallerID(<${MARLOW_SIPPHONE}>)
  exten => _1800.,2,Dial,SIP/*${EXTEN}@proxy01.sipphone.com
  exten => _1800.,3,Playback(invalid)
  exten => _1800.,4,Hangup
  exten => _1800.,103,Busy
-----------------------------------------------------------------------------------------------------------




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