I have the same problem. It works fine if the forward happened in<br><br><pre> # native SIP destinations are handled using our USRLOC DB<br> if (!lookup("location")) {<br> if(isflagset(2)) {<br> xlog("L_INFO", "Callee is Offline, call forward to Voice Mail - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
<br> # route to Asterisk Media Server<br> prefix("1");<br> rewritehostport("<a href="http://10.10.10.11:5060/" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">10.10.10.11:5060
</a>");<br> route(1);<br> } else {<br> sl_send_reply("404", "Not Found");<br><br> exit;<br> }</pre>May 18 09:06:21 localhost /usr/sbin/openser[24410]: Callee is Offline, call forward to Voice Mail - M=INVITE RURI=
<a href="mailto:sip:0280000000@10.10.1.2">sip:0280000000@10.10.1.2</a> F=<a href="mailto:sip:0299000000@10.10.1.2">sip:0299000000@10.10.1.2</a> T=<a href="mailto:sip:0280000000@10.10.1.2">sip:0280000000@10.10.1.2</a> IP=<a href="http://10.10.1.1">
10.10.1.1</a> ID=<a href="mailto:call-F11EC874-4CE7-2910-000A-3E6@10.10.1.1">call-F11EC874-4CE7-2910-000A-3E6@10.10.1.1</a><br><br>It is not working good in Failure_route<br><br><pre><br>failure_route[1] {<br> if (t_was_cancelled()) {
<br><br> xdbg("transaction was cancelled by UAC\n");<br> return;<br> }<br> xlog("L_INFO", "failure_route - call forward to Voice Mail - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");<br> # restore initial uri
<br> avp_pushto("$ruri", "i:10");<br> prefix("1");<br> # route to Asterisk Media Server<br><br> rewritehostport("<a href="http://10.10.10.11:5060/" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
10.10.10.11:5060</a>");<br> resetflag(2);<br> route(1);<br><br>}</pre><br><br>May 18 09:08:45 localhost /usr/sbin/openser[24414]: failure_route - call forward to Voice Mail - M=INVITE RURI=sip:0280000000@10.10.2.126:57042;rinstance=dbdab29df7aa260b F=
<a href="mailto:sip:0299000000@10.10.1.2">sip:0299000000@10.10.1.2</a> T=<a href="mailto:sip:0280000000@10.10.1.2">sip:0280000000@10.10.1.2</a> IP=<a href="http://10.10.1.1">10.10.1.1</a> ID=<a href="mailto:call-F17BFBB3-4FE7-2910-000C-3E8@10.10.1.1">
call-F17BFBB3-4FE7-2910-000C-3E8@10.10.1.1</a><br>May 18 09:08:59 localhost /usr/sbin/openser[24399]: ERROR:tm:t_forward_nonack: no branch for forwarding<br>May 18 09:08:59 localhost /usr/sbin/openser[24399]: ERROR:tm:w_t_relay: t_forward_nonack failed
<br>May 18 09:09:09 localhost /usr/sbin/openser[24399]: ERROR:tm:t_forward_nonack: failure to add branches<br><br>Anyone have an idea on where i have done wrong?<br><br>Regards,<br>Howard<br><br><br><div><span class="gmail_quote">
On 5/18/07, <b class="gmail_sendername">Bill Neely</b> <<a href="mailto:ceo@xantek.cc">ceo@xantek.cc</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
I am having a very similar problem. Using v1.2.0<br><br> Here is my route:<br>route[1] {<br><br>if(isflagset(2))<br> t_on_failure("2");<br><br><br> if (!t_relay()) {<br> sl_reply_error();<br> };<br>
exit;<br>}<br><br>failure_route[2]<br>{<br> if ( t_check_status("408"))<br><br> {<br> xlog("L_ERR","rrreeeeeeeeeeeeeeeeeecalling froute2 <$rm><$ru>\n");<br> avp_pushto("$ruri", "$avp(i:10)");
<br> prefix("777");<br> # route to Asterisk Media Server<br> rewritehostport("66.xxx.20.50:5060");<br> resetflag(2);<br><br> xlog("L_ERR","22222222222222222222calling froute2 <$rm><$ruri>\n");
<br> route(1);<br><br><br> }<br> exit;<br>}<br><br>Here is error message received:<br><br> 1(53165) rrreeeeeeeeeeeeeeeeeecalling froute2<br><INVITE><sip:1020101@67.188.xxx.188:35937;rinstance=e867c589f1896b12>
<br> 1(53165) 22222222222222222222calling froute2<br><INVITE><sip:7771020101@66.xxx.20.50:5060;rinstance=e867c589f1896b12><br> 1(53165) ERROR:tm:t_forward_nonack: no branch for forwarding<br> 1(53165) ERROR:tm:w_t_relay: t_forward_nonack failed
<br><br>Bogdan-Andrei Iancu wrote:<br>> Check with log/xlog prints if it gets to t_on_failure() and into<br>> failure route.<br>><br>> regards,<br>> Bogdan<br>><br>> Howard Tang wrote:<br>>> HI Bogdan,
<br>>><br>>> Thank you for your reply. I did that but i forget to include in this<br>>> email.<br>>><br>>><br>>> route[1] {<br>>> #check for nat flag<br>>> if (isflagset(2))
<br>>> {<br>>> fix_nated_contact();<br>>> use_media_proxy();<br>>> }<br>>><br>>> t_on_reply("1");<br>>> t_on_failure("1");
<br>>><br>>> # send it out now; use stateful forwarding as it works reliably<br>>> # even for UDP2TCP<br>>> xlog("L_INFO", "Request leaving server - M=$rm RURI=$ru F=$fu
<br>>> T=$tu IP=$si ID=$ci\n");<br>>> if (!t_relay()) {<br>>> if(isflagset(2))<br>>> end_media_session();<br>>> sl_reply_error();
<br>>> };<br>>> exit;<br>>> }<br>>><br>>> The voice mail work fine only when someone call in and the UA is<br>>> offline (not registered to the openser), if the UA is online, the
<br>>> call will ring the UA until the caller hang up.<br>>><br>>> I want to set up some sort of timer, i.e. 60 second and the call will<br>>> forwarded to the Voice mail.<br>>><br>>> Can you suggest me an idea on how i can make this happen please?
<br>>><br>>> Regards,<br>>> Howard<br>>><br>>><br>>><br>>> On 5/17/07, *Bogdan-Andrei Iancu* <<a href="mailto:bogdan@voice-system.ro">bogdan@voice-system.ro</a><br>>> <mailto:
<a href="mailto:bogdan@voice-system.ro">bogdan@voice-system.ro</a>>> wrote:<br>>><br>>> Hi Howard,<br>>><br>>> I guess you do not arm the failure route - use t_on_failure("1");
<br>>> before<br>>> relaying the request.<br>>><br>>> regards,<br>>> bogdan<br>>><br>>> Howard Tang wrote:<br>>> > Hi All,<br>>> ><br>
>> > I have followed a tutorial and set up Asterisk as a voice mail<br>>> server.<br>>> ><br>>> ><br>>><br>>> <a href="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</a><br>>><br>>> ><br>>><br>>> <<a href="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</a><br>>><br>>><br>>> <<a href="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
</a>>><br>>><br>>> ><br>>> > It works fine when the UA is offline. Now, I want a call<br>>> forwarded to<br>>> > the Voice mail server when there is no answer from the UA after 60
<br>>> > seconds(UA is registered on the openser).<br>>> ><br>>> > What should I do? Below is my config (copy from the above link).<br>>> ><br>>> ><br>>> > # requests for Media server
<br>>> > if(is_method("INVITE") && !has_totag() &&<br>>> uri=~"sip:\*9") {<br>>> > route(3);<br>>> > exit;
<br>>> > }<br>>> ><br>>> > # mark transaction if user is in voicemail group<br>>> ><br>>> > if(is_method("INVITE") && !has_totag()
<br>>> > && is_user_in("Request-URI","voicemail"))<br>>> > {<br>>> > xdbg("user [$ru] has voicemail redirection
<br>>> enabled\n");<br>>> ><br>>> > # backup R-URI<br>>> > avp_write("$ruri", "i:10");<br>>> > setflag(2);
<br>>> > };<br>>> ><br>>> > # native SIP destinations are handled using our<br>>> USRLOC DB<br>>> > if (!lookup("location")) {
<br>>> > if(isflagset(2)) {<br>>> ><br>>> > # route to Asterisk Media Server<br>>> > prefix("1");
<br>>> > rewritehostport("<a href="http://10.10.10.11:5060">10.10.10.11:5060</a><br>>> <<a href="http://10.10.10.11:5060">http://10.10.10.11:5060</a>> <<a href="http://10.10.10.11:5060">
http://10.10.10.11:5060</a>>");<br>>> > route(1);<br>>> > } else {<br>>> > sl_send_reply("404", "Not Found");
<br>>> ><br>>> > exit;<br>>> > }<br>>> > };<br>>> ><br>>> > # voicemail access
<br>>> > # - *98 - listen caller's voice messages, being prompted for pin<br>>> > # - *981 - listen voice messages, being promted for mailbox and<br>>> pin<br>>> > # - *98XXXX - leave voice message to XXXX
<br>>> ><br>>> > #<br>>> > route[3] {<br>>> > # direct voicemail<br>>> > if (uri =~ "sip:\*98@" ) {<br>>> > rewriteuser("1");
<br>>> > xdbg("voicemail access\n");<br>>> > } else if (uri =~ "sip:\*981@" ) {<br>>> ><br>>> > strip(4);<br>>> > rewriteuser("11");
<br>>> > } else if (uri =~ "sip:\*98.+@" ) {<br>>> > strip(3);<br>>> > prefix("1");<br>>> > } else {<br>>> > xlog("unknown media extension $rU\n");
<br>>> > sl_send_reply("404", "Unknown media service");<br>>> ><br>>> > exit;<br>>> > }<br>>> ><br>>> > # route to Asterisk Media Server
<br>>> > rewritehostport("<a href="http://10.10.10.11:5060">10.10.10.11:5060</a><br>>> <<a href="http://10.10.10.11:5060">http://10.10.10.11:5060</a>> < <a href="http://10.10.10.11:5060">
http://10.10.10.11:5060</a>>");<br>>> > route(1);<br>>> > }<br>>> ><br>>> > failure_route[1] {<br>>> > if (t_was_cancelled()) {<br>>> >
<br>>> > xdbg("transaction was cancelled by UAC\n");<br>>> > return;<br>>> > }<br>>> > # restore initial uri<br>>> > avp_pushto("$ruri", "i:10");
<br>>> > prefix("1");<br>>> > # route to Asterisk Media Server<br>>> ><br>>> > rewritehostport("<a href="http://10.10.10.11:5060">10.10.10.11:5060
</a><br>>> <<a href="http://10.10.10.11:5060">http://10.10.10.11:5060</a>> <<a href="http://10.10.10.11:5060">http://10.10.10.11:5060</a>>");<br>>> > resetflag(2);<br>>> > route(1);
<br>>> ><br>>> > }<br>>> ><br>>> ><br>>> ><br>>><br>>> ------------------------------------------------------------------------<br>>> >
<br>>> > _______________________________________________<br>>> > Users mailing list<br>>> > <a href="mailto:Users@openser.org">Users@openser.org</a> <mailto:<a href="mailto:Users@openser.org">
Users@openser.org</a>><br>>> > <a href="http://openser.org/cgi-bin/mailman/listinfo/users">http://openser.org/cgi-bin/mailman/listinfo/users</a><br>>> ><br>>><br>>><br>>><br>
>><br>>><br>>><br>>> --<br>>> Howard Tang<br>>> ICQ : 259083<br>>> MSN : <a href="mailto:howard615@hotmail.com">howard615@hotmail.com</a> <mailto:<a href="mailto:howard615@hotmail.com">
howard615@hotmail.com</a>><br>><br>><br>> _______________________________________________<br>> Users mailing list<br>> <a href="mailto:Users@openser.org">Users@openser.org</a><br>> <a href="http://openser.org/cgi-bin/mailman/listinfo/users">
http://openser.org/cgi-bin/mailman/listinfo/users</a><br><br>--<br>Bill Neely<br>Xantek, Inc.<br>1-866-553-3833<br>1-702-874-3833<br><br></blockquote></div><br><br clear="all"><br>-- <br>Howard Tang<br>ICQ : 259083<br>MSN :
<a href="mailto:howard615@hotmail.com">howard615@hotmail.com</a>