Guys Please Help,
I have tried all different options but i am still short of making and receiving calls.
This is what my i have:
sip.conf
context=GlobalPhone
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
allowexternaldomains=yes
allowguest=yes
allowsubscribe=yes
allowtransfer=yes
alwaysauthreject=no
autodomain=no
callevents=no
canreinvite=
checkmwi=10
compactheaders=no
defaultexpiry=120
domain=
dtmfmode=
dumphistory=no
externrefresh=10
fromdomain=
g726nonstandard=yes
jbenable=no
jbforce=no
jbimpl=
jblog=no
jbmaxsize=
jbresyncthreshold=
language=
maxcallbitrate=384
maxexpiry=3600
minexpiry=60
mohinterpret=default
mohsuggest=
nat=no
notifyringing=yes
pedantic=no
progressinband=never
promiscredir=no
realm=asterisk
recordhistory=no
registerattempts=0
registertimeout=20
relaxdtmf=no
rtpholdtimeout=
rtptimeout=
sendrpid=no
sipdebug=no
subscribecontext=
t1min=100
t38pt_udptl=no
tos_audio=none
tos_sip=none
tos_video=none
trustrpid=no
useragent=Asterisk PBX
usereqphone=no
videosupport=no
disallow=all
allow=undefined,ulaw,alaw,gsm
extensions.conf
exten=_91XXXXXXXXXX!,1,Macro(trunkdial-failover-0.3,${GlobalPhone}/${EXTEN:1},GlobalPhone,GlobalPhone)
exten=NXXNXXXXX,1,Macro(trunkdial-failover-0.3,${GlobalPhone}/${EXTEN:9},GlobalPhone,GlobalPhone)
Thanks again…just ask me what you need to know.