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<P><FONT SIZE=2>Someone can tell me if my sip.conf and my extension.conf are correctly for<BR>
setup Asterisk with SipPhone?<BR>
I'm a newbie about Asterisk, install it on a Suse 9.1 with a wildcard x100p<BR>
<BR>
*SIP.CONF*<BR>
[general]<BR>
port=5060<BR>
binaddr=0.0.0.0<BR>
disallow=all<BR>
allow=ulaw<BR>
allow=alaw<BR>
allow=gsm<BR>
register => <my number Sipphone>:<pwdSipPhone>@proxy01.sipphone.com/dialout<BR>
<BR>
<BR>
;be sure to set your external ip if you're behind a router.<BR>
;Especially if you want to use sipphone minutes<BR>
externip=xxx.xxx.xxx.xxx<BR>
<BR>
<BR>
;uncomment this line as it should be in your default sip.conf<BR>
localnet=192.168.1.0/255.255.255.0; All RFC 1918 addresses are local networks<BR>
<BR>
<BR>
;Note: the name is [proxy01.sipphone.com] so that incoming calls work<BR>
correctly<BR>
[proxy01.sipphone.com]<BR>
type=friend<BR>
;secret=2128<BR>
username=mynumbersipphone<BR>
host=proxy01.sipphone.com<BR>
dtmfmode=inband<BR>
context=dialout ;or your context which includes the dialing rules<BR>
nat=yes<BR>
qualify=no<BR>
reinvite=no<BR>
canreinvite=no<BR>
<BR>
*EXTENSIONS.CONF*<BR>
[general]<BR>
static=yes<BR>
writeprotect=yes<BR>
<BR>
[globals]<BR>
MD110=Zap/g2<BR>
<BR>
SIPPHONEUSERID=mynumbersipphone<BR>
MYNAME=Mauro<BR>
[dialout]<BR>
<BR>
<BR>
include => sip-forced<BR>
include => from-sipphone<BR>
<BR>
<BR>
; Check to see if the called number starts with a "6" and<BR>
; if so, set the call parameters and bounce the call to the<BR>
; SipPhone.com SIP server.<BR>
;<BR>
; NOTE: Calls to unknown users will result in "invalid extension"<BR>
; message being played.<BR>
;<BR>
[sip-forced]<BR>
<BR>
exten => _6.,1,SetCallerID(${SIPPHONEUSERID})<BR>
exten => _6.,2,SetCIDName(${MYNAME})<BR>
exten => _6.,3,Dial(SIP/${EXTEN:1}@proxy01.sipphone.com)<BR>
exten => _6.,4,Playback(invalid)<BR>
exten => _6.,5,Hangup<BR>
<BR>
<BR>
; To receive calls inbound from SipPhone.com, we set the extension<BR>
; to our SipPhone user id, in this case from the SIPPHONEUSERID variable<BR>
; Changing the "Dial"<BR>
; directive to something like this:<BR>
; Dial(${PHONES1}&${PHONES2},15,Ttm)<BR>
; would cause both lines to ring<BR>
;<BR>
<BR>
[from-sipphone]<BR>
exten => ${SIPPHONEUSERID},1,Dial(${PHONES1},30,Ttm)<BR>
exten => ${SIPPHONEUSERID},2,Voicemail2(u${PHONES1VM})<BR>
exten => ${SIPPHONEUSERID},3,Hangup<BR>
<BR>
This are my complete file sip and extensions..<BR>
<BR>
Thanks..<BR>
Mauro<BR>
P.s. My english is very bad:( if ther is something italian pls mail me:D</FONT>
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