hi all<br>Sorry for delay<br><br>these are my few lines to include in the routing logic to manage OpenSips + Asterisk IVR<br><br>On OpenSips Box<br><br>if($rU=="<aexten>){<br> rewriteuri("sip:<aexten>@<asterisk_ip>:<asterisk_port>");<br> t_relay();<br> resetflag(IDN); # Reset flag used to get in this subroutine<br> }<br><br>On Asterisk Box<br><br>[inbound-context]<br>include => exten_context<br><br>[exten_context]<br>exten => aexten,1,Answer()<br>exten => aexten,n,Background(/ivr/ivr_file)<br>exten => aexten,n,WaitExten(10)<br>exten => aexten,n,NoOp()<br>exten => aexten,n,Return()<br>exten => t,1,Hangup()<br><br>exten => 1,1,Dial(SIP/<outbound_trunk>/<called_number>,20)<br>exten => 2,1,Dial(SIP/<outbound_trunk>/<called_number_2>,20)<br><br>
<br>Hope this can help<br><br><br>
----Messaggio originale----<br>Da: mail@sathees.co.uk<br>Data: 14-mar-2015 18.11<br>A: <users@lists.opensips.org><br>Ogg: Re: [OpenSIPS-Users] OPENSIPS + IVR CALL CONTROL<br><br>Hello again Danilo,<br><br>Thank you for the quick replay.<br><br>I have asterisk server running at public IP. <br><br>I have to use IVR, Voicemail, on hold message and incoming DDIs. All<br>incoming DDI send direct to asterisk IP.<br><br> Some DDI will play welcome message while phone rings, others will ring<br>group and after certain time it will go direct to closed message. All these<br>functions are working with asterisk right now. I’m getting high-level sip<br>flood attack. Now I’m trying to secure server with OpenSIPS. That’s why if<br>I see your scripts it will help me understand more. Looking forward to your<br>email on Monday<br><br>Many thanks<br>sathees<br><br><br><br><br>--<br>View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/OPENSIPS-IVR-CALL-CONTROL-tp7595634p7595890.html<br>Sent from the OpenSIPS - Users mailing list archive at Nabble.com.<br><br>_______________________________________________<br>Users mailing list<br>Users@lists.opensips.org<br>http://lists.opensips.org/cgi-bin/mailman/listinfo/users<br><br><br>