Penguin
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An Example setup for connecting from Asterisk to sipphone.com.

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.

extensions.conf

[globals]
.
.
SIPPHONEUSERID=1747xxxXXXX
MYNAME=Bob
[[dialout]
.
.
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
; SipPhone.com SIP server.
;
; NOTE: Calls to unknown users will result in "invalid extension"
; message being played.
;
[[sip-forced]

exten => _6.,1,SetCallerID(${SIPPHONEUSERID})
exten => _6.,2,SetCIDName(${MYNAME})
exten => _6.,3,Dial(SIP/${EXTEN:1}@proxy01.sipphone.com)
exten => _6.,4,Playback(invalid)
exten => _6.,5,Hangup


; To receive calls inbound from SipPhone.com, 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
;

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

sip.conf

[[general]
.
.
.
register=<YourSIPPhoneID>:<YourSIPPhonePassword>@proxy01.sipphone.com/<extension>
.
.
;be sure to set your external ip if you're behind a router.
;Especially if you want to use sipphone minutes
externip=xxx.xxx.xxx.xxx
.
.
;uncomment this line as it should be in your default sip.conf
localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks



;Note: the name is [proxy01.sipphone.com] so that incoming calls work correctly
[proxy01.sipphone.com]
type=friend
secret=<YourSIPPhonePassword>
username=<YourSIPPhoneID>
host=proxy01.sipphone.com
dtmfmode=inband
context=dialout ;or your context which includes the dialing rules
nat=yes
qualify=no
reinvite=no
canreinvite=no
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  • TonyAllen
  • AdamWentworth
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