Re: [Users] Call forward when no answer

I am having a very similar problem. Using v1.2.0

Here is my route: route[1] {

if(isflagset(2)) t_on_failure("2");

if (!t_relay()) { sl_reply_error(); }; exit; }

failure_route[2] { if ( t_check_status("408"))

{ xlog("L_ERR","rrreeeeeeeeeeeeeeeeeecalling froute2 <$rm><$ru>\n"); avp_pushto("$ruri", "$avp(i:10)"); prefix("777"); # route to Asterisk Media Server rewritehostport("66.xxx.20.50:5060"); resetflag(2);

xlog("L_ERR","22222222222222222222calling froute2 <$rm><$ruri>\n"); route(1);

} exit; }

Here is error message received:

1(53165) rrreeeeeeeeeeeeeeeeeecalling froute2 <INVITE>sip:1020101@67.188.xxx.188:35937;rinstance=e867c589f1896b12 1(53165) 22222222222222222222calling froute2 <INVITE>sip:7771020101@66.xxx.20.50:5060;rinstance=e867c589f1896b12 1(53165) ERROR:tm:t_forward_nonack: no branch for forwarding 1(53165) ERROR:tm:w_t_relay: t_forward_nonack failed

Bogdan-Andrei Iancu wrote:

Check with log/xlog prints if it gets to t_on_failure() and into failure route.

regards, Bogdan

Howard Tang wrote:

...

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 mailto: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...

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> <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 $rU\n"); > sl_send_reply("404", "Unknown media service"); > > exit; > } > > # route to Asterisk Media Server > rewritehostport("10.10.10.11:5060 http://10.10.10.11:5060 < http://10.10.10.11:5060%3E"); > route(1); > } > > failure_route[1] { > if (t_was_cancelled()) { > > xdbg("transaction was cancelled by UAC\n"); > return; > } > # restore initial uri > avp_pushto("$ruri", "i:10"); > prefix("1"); > # route to Asterisk Media Server > > rewritehostport("10.10.10.11:5060 http://10.10.10.11:5060 http://10.10.10.11:5060"); > resetflag(2); > route(1); > > } > > >

---

>
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>

-- Howard Tang ICQ : 259083 MSN : howard615@hotmail.com mailto:howard615@hotmail.com

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