hi all
Sorry for delay
these are my few lines to include in the routing logic to manage OpenSips +
Asterisk IVR
On OpenSips Box
if($rU=="<aexten>){
rewriteuri("sip:<aexten>@<asterisk_ip>:<asterisk_port>");
t_relay();
resetflag(IDN); # Reset flag used to get in this subroutine
}
On Asterisk Box
[inbound-context]
include => exten_context
[exten_context]
exten => aexten,1,Answer()
exten => aexten,n,Background(/ivr/ivr_file)
exten => aexten,n,WaitExten(10)
exten => aexten,n,NoOp()
exten => aexten,n,Return()
exten => t,1,Hangup()
exten => 1,1,Dial(SIP/<outbound_trunk>/<called_number>,20)
exten => 2,1,Dial(SIP/<outbound_trunk>/<called_number_2>,20)
Hope this can help
----Messaggio originale----
Da: [email protected]
Data: 14-mar-2015 18.11
A: <[email protected]>
Ogg: Re: [OpenSIPS-Users] OPENSIPS + IVR CALL CONTROL
Hello again Danilo,
Thank you for the quick replay.
I have asterisk server running at public IP.
I have to use IVR, Voicemail, on hold message and incoming DDIs. All
incoming DDI send direct to asterisk IP.
Some DDI will play welcome message while phone rings, others will ring
group and after certain time it will go direct to closed message. All these
functions are working with asterisk right now. I’m getting high-level sip
flood attack. Now I’m trying to secure server with OpenSIPS. That’s why if
I see your scripts it will help me understand more. Looking forward to your
email on Monday
Many thanks
sathees
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