Hello forumers. Im kinda new on this. I have registered a voxalot and gizmo accounts to my asterisk server. all outgoing calls to voxalot echo test number and gizmo “tell me” are ok.
CUrrent Im setting up for Incoming for both providers but the calls are dropped.
What i want to achieve is that all incoming calls either from voxalot or sipphone will be transferred to extension 5000, if not answered, it will goes into voicemailbox.
attached here are my sip.conf and extensions.conf
sip.conf
[general]
register => 17474XXXXXX:XXXXXX@proxy01.sipphone.com
register => 20XXXX:PASSWORD@voxalot.com
context=phones
allowoverlap=yes
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
localnet=192.168.0.0/255.255.0.0
sets
type=friend
host=dynamic
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
nat=no
dtmfmode=rfc2833
[proxy01.sipphone.com]
type=friend
context=to_gizmo
host=proxy01.sipphone.com
secret=PASSWORD
defaultuser=17474XXXXXX
fromdomain=proxy01.sipphone.com
fromuser=17474XXXXXX
qualify=yes
[voxalot.com]
defaultuser = 20XXXX
fromuser = 20XXXX
callerid = VOX1 <20XXXX>
secret = PASSWORD
host = us.voxalot.com
fromdomain = voxalot.com
context=to_voxalot
type=friend
qualify=yes
my extensions.conf look like this
[globals]
exten => s,1,Set(CallerIDString=${CALLERID(all)})
exten => s,1,Set(CALLERID(name)=Asterisk PBX)
[default]
exten => s,1,noop(got a call)
exten => s,2,ExecIf($[ “${CALLERID(num)}”="" ]?SetCallerPres(unavailable))
exten => s,3,ExecIf($[ “${CALLERID(num)}”="" ]?Set(CALLERID(all)=unknown <0000000>))
exten => s,4,Dial(SIP/5000,20,t)
exten => s,5,Voicemail(5000) ; go to Voicemail2 if phone is "U"nanswered
exten => s,6,Hangup
[to_gizmo]
exten => _91747.,1,Dial(SIP/${EXTEN:1}@proxy01.sipphone.com,20,r)
exten => _91747.,2,Congestion(5)
exten => _91222.,1,Dial(SIP/${EXTEN:1}@proxy01.sipphone.com,20,r)
exten => _91222.,2,Congestion(5)
exten => _49X.,1,Dial(SIP/${EXTEN:2}@proxy01.sipphone.com,20,r)
exten => _9411,1,Dial(SIP/${EXTEN:1}@proxy01.sipphone.com,20,r)
exten => _9.,1,Dial(SIP/${EXTEN:1}@proxy01.sipphone.com,20,r)
[from_gizmo]
exten => s,1,Set(CallerIDString=${CALLERID(all)})
exten => s,2,Dial(SIP/5000,20,t)
exten => s,3,Voicemail(5000) ; go to Voicemail2 if phone is "U"nanswered
exten => s,4,Hangup ; and then hangup
[to_voxalot]
exten => _8.,1,Dial(SIP/${EXTEN:1}@voxalot.com,20,r)
[internal]
exten => *97,1,VoiceMailMain ; go to Voicemail
exten => _5XXX,1,NoOp()
exten => _5XXX,n,Dial(SIP/${EXTEN},30)
exten => _5XXX,n,Playback(tt-allbusy)
exten => _5XXX,n,Wait(1)
exten => _5XXX,n,Voicemail(${EXTEN}) ; go to Voicemail2 if phone is "U"nanswered
exten => _5XXX,n,Hangup()
[phones]
include => internal
include => to_gizmo
include => from_gizmo
include => to_voxalot
include => default
the results when calls come in from voxalot or gizmo
== Using SIP RTP CoS mark 5
== Using SIP RTP CoS mark 5
== Using SIP RTP CoS mark 5
i am using asterisk version 1.6.0.9