route[1] {
       #check for nat flag
       if (isflagset(2))
       {
               fix_nated_contact();
               use_media_proxy();
       }

       # send it out now; use stateful forwarding as it works reliably
       # even for UDP2TCP
       xlog("L_INFO", "Request leaving server - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
       if (!t_relay()) {
               if(isflagset(2))
                       end_media_session();
               sl_reply_error();
       };
       exit;
}

On 5/17/07, Bogdan-Andrei Iancu <bogdan@voice-system.ro> wrote:

Hi Howard,

I guess you do not arm the failure route - use t_on_failure("1"); before
relaying the request.

regards,
bogdan

Howard Tang wrote:
> Hi All,
>
> I have followed a tutorial and set up Asterisk as a voice mail server.
>
> http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+With+OpenSER
> <http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+With+OpenSER >
>
> It works fine when the UA is offline. Now, I want a call forwarded to
> the Voice mail server when there is no answer from the UA after 60
> seconds(UA is registered on the openser).
>
> What should I do? Below is my config (copy from the above link).
>
>
>               # requests for Media server
>               if(is_method("INVITE") && !has_totag() && uri=~"sip:\*9") {
>                       route(3);
>                       exit;
>               }
>
>               # mark transaction if user is in voicemail group
>
>               if(is_method("INVITE") && !has_totag()
>                       && is_user_in("Request-URI","voicemail"))
>               {
>                       xdbg("user [$ru] has voicemail redirection enabled\n");
>
>                       # backup R-URI
>                       avp_write("$ruri", "i:10");
>                       setflag(2);
>               };
>
>               # native SIP destinations are handled using our USRLOC DB
>               if (!lookup("location")) {
>                       if(isflagset(2)) {
>
>                               # route to Asterisk Media Server
>                               prefix("1");
>                               rewritehostport("10.10.10.11:5060 <http://10.10.10.11:5060>");
>                               route(1);
>                       } else {
>                               sl_send_reply("404", "Not Found");
>
>                               exit;
>                       }
>               };
>
> # voicemail access
> # - *98 - listen caller's voice messages, being prompted for pin
> # - *981 - listen voice messages, being promted for mailbox and pin
> # - *98XXXX - leave voice message to XXXX
>
> #
> route[3] {
>       # direct voicemail
>       if (uri =~ "sip:\*98@" ) {
>               rewriteuser("1");
>               xdbg("voicemail access\n");
>       } else if (uri =~ "sip:\*981@" ) {
>
>               strip(4);
>               rewriteuser("11");
>       } else if (uri =~ "sip:\*98.+@" ) {
>               strip(3);
>               prefix("1");
>       } else {
>               xlog("unknown media extension $rU\n");
>               sl_send_reply("404", "Unknown media service");
>
>               exit;
>       }
>
>       # route to Asterisk Media Server
>       rewritehostport("10.10.10.11:5060 < http://10.10.10.11:5060>");
>       route(1);
> }
>
> failure_route[1] {
>       if (t_was_cancelled()) {
>
>               xdbg("transaction was cancelled by UAC\n");
>               return;
>       }
>       # restore initial uri
>       avp_pushto("$ruri", "i:10");
>       prefix("1");
>       # route to Asterisk Media Server
>
>       rewritehostport("10.10.10.11:5060 <http://10.10.10.11:5060>");
>       resetflag(2);
>       route(1);
>
> }
>
>
> ------------------------------------------------------------------------
>
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>