thanks williamconley u gave me a good detailled help.
this is my conf
1/ gotoautodial 3.0 CE
2/ Asterisk 1.4.39.1-vici.go
3/ i686 running Linux , single server
4/installation from scratch
[voip1]
srvlookup=yes
disallow=all
allow=g729
type=friend
authusername=XXXX72
username=XXXX72
secret=XXX18
host=sip.test.com
fromdomain=sip.test.com
authuser=XXXX72
fromuser=XXXX72
dtmfmode=rfc2833
qualify=1000
[voip2]
srvlookup=yes
disallow=all
allow=g729
type=friend
authusername=XXXX70
username=XXXX70
secret=XXX140
host=sip.test2.com
fromdomain=sip.test2.com
authuser=XXXX70
fromuser=XXXX70
dtmfmode=rfc2833
qualify=1000
#phones#
[8001]
username=8001
secret=goautodial
accountcode=8001
callerid="8001" <0000000000>
mailbox=8001
type=friend
host=dynamic
canreinvite=no
context=default
disallow=all
allow=g729
deny=0.0.0.0/0.0.0.0
permit=10.0.0.0/0.0.0.0
#dial plan#
; voip1
exten => _[89]ZXXXXXXX,1,AGI(
agi://127.0.0.1:4577/call_log)
exten => _[89]ZXXXXXXX,2,Set(CALLFILENAME=${CALLERID}-${EXTEN}-${STRFTIME(${EPOCH},,$
exten => _[89]ZXXXXXXX,3,Monitor(wav,${CALLFILENAME},m)
exten => _[89]ZXXXXXXX,4,Dial(SIP/${EXTEN}@voip1,29,tToR)
exten => _[89]ZXXXXXXX,5,Hangup()
;voip2
exten => _[6]XXXXXXXX,1,AGI(
agi://127.0.0.1:4577/call_log)
exten => _[6]XXXXXXXX,2,Set(CALLFILENAME=${CALLERID}-${EXTEN}-${STRFTIME(${EPOCH},,%$
exten => _[6]XXXXXXXX,3,Monitor(wav,${CALLFILENAME},m)
exten => _[6]XXXXXXXX,4,Dial(SIP/${EXTEN}@voip2,29,tToR)
exten => _[6]XXXXXXXX,5,Hangup()
so what i understand from u is to change
exten => _[6]XXXXXXXX,4,Dial(SIP/${EXTEN}@voip2,29,tToR)
To
exten => _33[6]XXXXXXXX,4,Dial(SIP/${EXTEN:1}@voip2,29,tToR)
33 as prefix, and why in interface web i cant hear ringing ??