Re: [Users] Call forward when no answer

HI Bogdan,

Thank you for your reply. I did that but i forget to include in this email.

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

t_on_reply("1"); t_on_failure("1");

# 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; } The voice mail work fine only when someone call in and the UA is offline (not registered to the openser), if the UA is online, the call will ring the UA until the caller hang up.

I want to set up some sort of timer, i.e. 60 second and the call will forwarded to the Voice mail.

Can you suggest me an idea on how i can make this happen please?

Regards, Howard

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+Ope...

...

<

http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+With+Ope...

...

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%3E");

...

                          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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