I success to register trank from Pbx1 to Pbx2 , and i received calls from num 747104354 but there is no voice
this is my lap please help me how can i fixed to transfer voice .
My lab :
(((((pbx1))))))
my sip.conf
[general]
bindport=5060
context=default ; Default context for incoming calls. Defaults to ‘default’
allowoverlap=yes ; Disable overlap dialing support. (Default is yes)
udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
tcpenable=no ; Enable server for incoming TCP connections (default is no)
;tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
transport=udp ; Set the default transports. The order determines the primary default transport.
qualify=yes
nat=force_rport,comedia
[Sipprovider]
host=80.179.122.145
type=friend
qualify=yes
context=from-pstn
canreinvite=no
insecure=very
nat=yes
disallow=all
allowguest=no
alwaysauthreject=yes
allow=alaw&ulaw&gsm
[SIPmylap]
username=SIPmylap
type=friend
secret=Pp123456!
;qualify=yes
host=52.25.25.25
context=mylap
disallow=all
allow=ulaw
allow=alaw
allow=gsm
[216]
type=friend
context=phones
allow=ulaw
allow=alaw
allow=gsm
secret=12345678
host=dynamic
(((((((extension.conf))))))))
[from-internal]
exten => 747104350,1,NoOp(Call received)
same => n,Set(Monitor_File_Name=/var/www/html/voip/records/{EXTEN}-{UNIQUEID})
same => n,MixMonitor(${Monitor_File_Name}.wav,a)
same => n,Goto(phones,216,1)
exten => 747104354,1,SetCallerID(747104354)
same => Dial(SIP/${EXTEN}@SIPmylap)
include => outgoing
include => phones
[phones]
exten => 216,1,NoOp(First Line)
same => n,NoOp(Secend Line)
same => n,Set(Monitor_File_Name=/var/spool/asterisk/monitor/{UNIQUEID})
same => n,MixMonitor({Monitor_File_Name}.wav,a)
same => n,Dial(SIP/216)
same => n,Hangup
include => outgoing
include => from-internal
[outgoing]
exten => _X.,1,Answer
exten => _X.,n,Ringing()
exten => _X.,n,Wait(2)
exten => _X.,n,Set(CALLERID(num)=747104354)
exten => _X.,n,Set(Monitor_File_Name=/var/spool/asterisk/monitor/{EXTEN}-{UNIQUEID})
exten => _X.,n,MixMonitor({Monitor_File_Name}.wav,a)
exten => _X.,n,Dial(SIP/Sipprovider/{EXTEN})
exten => _X.,n,Hangup
(((((PBX2))))))
(((((((Sip.conf))))))
[general]
bindport=5060
context=default ; Default context for incoming calls. Defaults to ‘default’
allowoverlap=yes ; Disable overlap dialing support. (Default is yes)
udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
tcpenable=no ; Enable server for incoming TCP connections (default is no)
;tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
transport=udp ; Set the default transports. The order determines the primary default transport.
qualify=yes
nat=force_rport,comedia
[SIPmylap]
username=SIPmylap
type=friend
secret=Pp123456!
;qualify=yes
host=52.25.35.35
context=mylap
disallow=all
allow=ulaw
allow=alaw
allow=gsm
[2116]
type=friend
context=phones
allow=ulaw
allow=alaw
allow=gsm
secret=12345678
host=dynamic
(((((((extension.conf))))))))
[from-internal]
exten => 747104354,1,NoOp(Call received)
same => n,Set(Monitor_File_Name=/var/www/html/voip/records/{EXTEN}-{UNIQUEID})
same => n,MixMonitor(${Monitor_File_Name}.wav,a)
same => n,Goto(phones,2116,1)
include => outgoing
include => phones
[phones]
exten => 2116,1,NoOp(First Line)
same => n,NoOp(Secend Line)
same => n,Set(Monitor_File_Name=/var/spool/asterisk/monitor/{UNIQUEID})
same => n,MixMonitor({Monitor_File_Name}.wav,a)
same => n,Dial(SIP/2116)
same => n,Hangup
include => outgoing
include => from-internal
[outgoing]
exten => _X.,1,Answer
exten => _X.,n,Ringing()
exten => _X.,n,Wait(2)
exten => _X.,n,Set(CALLERID(num)=747104354)
exten => _X.,n,Set(Monitor_File_Name=/var/spool/asterisk/monitor/{EXTEN}-{UNIQUEID})
exten => _X.,n,MixMonitor({Monitor_File_Name}.wav,a)
exten => _X.,n,Dial(SIP/SIPmylap/{EXTEN})
exten => _X.,n,Hangup