An Example setup for connecting from Asterisk to

Note: If you are behind a router/firewall, you'll want to forward the ports 5004, 5060, and 69 to your local ip with the asterisk server on it.


include => sip-forced
include => from-sipphone

; Check to see if the called number starts with a "6" and
; if so, set the call parameters and bounce the call to the
; SIP server.
; NOTE: Calls to unknown users will result in "invalid extension"
; message being played.

exten => _6.,1,SetCallerID(${SIPPHONEUSERID})
exten => _6.,2,SetCIDName(${MYNAME})
exten => _6.,3,Dial(SIP/${EXTEN:1}
exten => _6.,4,Playback(invalid)
exten => _6.,5,Hangup

; To receive calls inbound from, we set the extension
; to our SipPhone user id, in this case from the SIPPHONEUSERID variable
; Changing the "Dial"
; directive to something like this:
; Dial(${PHONES1}&${PHONES2},15,Ttm)
; would cause both lines to ring

exten => ${SIPPHONEUSERID},1,Dial(${PHONES1},30,Ttm)
exten => ${SIPPHONEUSERID},2,Voicemail2(u${PHONES1VM})
exten => ${SIPPHONEUSERID},3,Hangup


;be sure to set your external ip if you're behind a router.
;Especially if you want to use sipphone minutes
;uncomment this line as it should be in your default sip.conf
localnet=; All RFC 1918 addresses are local networks

;Note: the name is [] so that incoming calls work correctly
context=dialout ;or your context which includes the dialing rules
canreinvite=no ; I was unable to dial out to PSTN using SIP Minutes without this line, Failed INVITE :)
The following authors of this page have not agreed to the WlugWikiLicense. As such copyright to all content on this page is retained by the original authors.
  • TonyAllen
  • AdamWentworth
The following authors of this page have agreed to the WlugWikiLicense.

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