An Example setup for connecting from Asterisk to sipphone.com

__In extensions.conf [Globals?
.
.
SIPPHONEUSERID=17473863406

[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}@sipphone)
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


__In sip.conf

[general?
.
.
.
register=<YourSIPPhoneID>:<YourSIPPhonePassword>@proxy01.sipphone.com

[sipphone?
type=friend
secret=<YourSIPPhonePassword>
username=<YourSIPPhoneID>
host=proxy01.sipphone.com
dtmfmode=inband
context=home
nat=yes
reinvite=no
canreinvite=no
disallow=all
allow=all