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

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

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