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...
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"); 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 $rU\n"); sl_send_reply("404", "Unknown media service"); exit; }
# route to Asterisk Media Server rewritehostport("10.10.10.11:5060"); route(1); }
failure_route[1] { if (t_was_cancelled()) { xdbg("transaction was cancelled by UAC\n"); return; } # restore initial uri avp_pushto("$ruri", "i:10"); prefix("1"); # route to Asterisk Media Server rewritehostport("10.10.10.11:5060"); resetflag(2); route(1);
}
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+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);
}
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
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 [$ru] has voicemail redirectionenabled\n");
# backup R-URI avp_write("$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 <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);}
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
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=$fuT=$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+OpenSER > <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> <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 $rU\n"); > sl_send_reply("404", "Unknown media service"); > > exit; > } > > # route to Asterisk Media Server > rewritehostport("10.10.10.11:5060 <http://10.10.10.11:5060> < http://10.10.10.11:5060>"); > route(1); > } > > failure_route[1] { > if (t_was_cancelled()) { > > xdbg("transaction was cancelled by UAC\n"); > return; > } > # restore initial uri > avp_pushto("$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); > > } > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users@openser.org <mailto:Users@openser.org> > http://openser.org/cgi-bin/mailman/listinfo/users >-- Howard Tang ICQ : 259083 MSN : howard615@hotmail.com mailto:howard615@hotmail.com
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=$fuT=$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 andpin > # - *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); > > } > > >
> > _______________________________________________ > Users mailing list > Users@openser.org <mailto:Users@openser.org> > http://openser.org/cgi-bin/mailman/listinfo/users >-- Howard Tang ICQ : 259083 MSN : howard615@hotmail.com mailto:howard615@hotmail.com
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
I have the same problem. It works fine if the forward happened in
# native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { if(isflagset(2)) { xlog("L_INFO", "Callee is Offline, call forward to Voice Mail - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # route to Asterisk Media Server prefix("1"); rewritehostport("10.10.10.11:5060"); route(1); } else { sl_send_reply("404", "Not Found");
exit; }
May 18 09:06:21 localhost /usr/sbin/openser[24410]: Callee is Offline, call forward to Voice Mail - M=INVITE RURI=sip:0280000000@10.10.1.2 F= sip:0299000000@10.10.1.2 T=sip:0280000000@10.10.1.2 IP=10.10.1.1 ID= call-F11EC874-4CE7-2910-000A-3E6@10.10.1.1
It is not working good in Failure_route
failure_route[1] { if (t_was_cancelled()) {
xdbg("transaction was cancelled by UAC\n"); return; } xlog("L_INFO", "failure_route - call forward to Voice Mail - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # restore initial uri avp_pushto("$ruri", "i:10"); prefix("1"); # route to Asterisk Media Server
rewritehostport("10.10.10.11:5060"); resetflag(2); route(1);
}
May 18 09:08:45 localhost /usr/sbin/openser[24414]: failure_route - call forward to Voice Mail - M=INVITE RURI=sip:0280000000@10.10.2.126:57042;rinstance=dbdab29df7aa260b F=sip:0299000000@10.10.1.2 T=sip:0280000000@10.10.1.2 IP=10.10.1.1 ID= call-F17BFBB3-4FE7-2910-000C-3E8@10.10.1.1 May 18 09:08:59 localhost /usr/sbin/openser[24399]: ERROR:tm:t_forward_nonack: no branch for forwarding May 18 09:08:59 localhost /usr/sbin/openser[24399]: ERROR:tm:w_t_relay: t_forward_nonack failed May 18 09:09:09 localhost /usr/sbin/openser[24399]: ERROR:tm:t_forward_nonack: failure to add branches
Anyone have an idea on where i have done wrong?
Regards, Howard
On 5/18/07, Bill Neely ceo@xantek.cc wrote:
I am having a very similar problem. Using v1.2.0
Here is my route: route[1] {
if(isflagset(2)) t_on_failure("2");
if (!t_relay()) { sl_reply_error(); }; exit;}
failure_route[2] { if ( t_check_status("408"))
{ xlog("L_ERR","rrreeeeeeeeeeeeeeeeeecalling froute2 <$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=$ru F=$fuT=$tu IP=$si ID=$ci\n"); if (!t_relay()) { if(isflagset(2)) end_media_session(); sl_reply_error(); }; exit; }
The voice mail work fine only when someone call in and the UA is offline (not registered to the openser), if the UA is online, the call will ring the UA until the caller hang up.
I want to set up some sort of timer, i.e. 60 second and the call will forwarded to the Voice mail.
Can you suggest me an idea on how i can make this happen please?
Regards, Howard
On 5/17/07, *Bogdan-Andrei Iancu* <bogdan@voice-system.ro mailto:bogdan@voice-system.ro> wrote:
Hi Howard, I guess you do not arm the failure route - use t_on_failure("1"); before relaying the request. regards, bogdan Howard Tang wrote: > Hi All, > > I have followed a tutorial and set up Asterisk as a voice mail server. > >http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+With+Ope...
><
http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+With+Ope...
<
http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+With+Ope...
> > It works fine when the UA is offline. Now, I want a call forwarded to > the Voice mail server when there is no answer from the UA after60
> seconds(UA is registered on the openser). > > What should I do? Below is my config (copy from the above link). > > > # requests for Media server > if(is_method("INVITE") && !has_totag() && uri=~"sip:\*9") { > route(3); > exit; > } > > # mark transaction if user is in voicemail group > > if(is_method("INVITE") && !has_totag() > && is_user_in("Request-URI","voicemail")) > { > xdbg("user [$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 andpin > # - *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); > > } > > >
> > _______________________________________________ > Users mailing list > Users@openser.org <mailto:Users@openser.org> > http://openser.org/cgi-bin/mailman/listinfo/users >-- Howard Tang ICQ : 259083 MSN : howard615@hotmail.com mailto:howard615@hotmail.com
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
-- Bill Neely Xantek, Inc. 1-866-553-3833 1-702-874-3833
HI All,
Problem is fixed by adding append_branch(); in the failure_route
failure_route[1] { if (t_was_cancelled()) {
xdbg("transaction was cancelled by UAC\n"); return; } xlog("L_INFO", "failure_route - call forward to Voice Mail - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # restore initial uri
avp_pushto("$ruri", "i:10"); prefix("1"); # route to Asterisk Media Server
rewritehostport(" 10.10.10.11:5060"); resetflag(2); append_branch(); route(1);
}
Regards, Howard
On 5/18/07, Howard Tang howard615@gmail.com wrote:
I have the same problem. It works fine if the forward happened in
# native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { if(isflagset(2)) { xlog("L_INFO", "Callee is Offline, call forward to Voice Mail - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
# route to Asterisk Media Server prefix("1"); rewritehostport("10.10.10.11:5060"); route(1); } else { sl_send_reply("404", "Not Found");
exit; }May 18 09:06:21 localhost /usr/sbin/openser[24410]: Callee is Offline, call forward to Voice Mail - M=INVITE RURI= sip:0280000000@10.10.1.2 F= sip:0299000000@10.10.1.2 T=sip:0280000000@10.10.1.2 IP= 10.10.1.1 ID= call-F11EC874-4CE7-2910-000A-3E6@10.10.1.1
It is not working good in Failure_route
failure_route[1] { if (t_was_cancelled()) {
xdbg("transaction was cancelled by UAC\n"); return;} xlog("L_INFO", "failure_route - call forward to Voice Mail - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # restore initial uri
avp_pushto("$ruri", "i:10"); prefix("1"); # route to Asterisk Media Server
rewritehostport(" 10.10.10.11:5060"); resetflag(2); route(1);
}
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= sip:0299000000@10.10.1.2 T=sip:0280000000@10.10.1.2 IP=10.10.1.1 ID=call-F17BFBB3-4FE7-2910-000C-3E8@10.10.1.1 May 18 09:08:59 localhost /usr/sbin/openser[24399]: ERROR:tm:t_forward_nonack: no branch for forwarding May 18 09:08:59 localhost /usr/sbin/openser[24399]: ERROR:tm:w_t_relay: t_forward_nonack failed May 18 09:09:09 localhost /usr/sbin/openser[24399]: ERROR:tm:t_forward_nonack: failure to add branches
Anyone have an idea on where i have done wrong?
Regards, Howard
On 5/18/07, Bill Neely ceo@xantek.cc wrote:
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 worksreliably
# even for UDP2TCP xlog("L_INFO", "Request leaving server - M=$rm RURI=$ru F=$fuT=$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+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 after60
> seconds(UA is registered on the openser). > > What should I do? Below is my config (copy from the abovelink).
> > > # 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 [$ru] has voicemailredirection
enabled\n"); > > # backup R-URI > avp_write("$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", "NotFound");
> > exit; > } > }; > > # voicemail access > # - *98 - listen caller's voice messages, being prompted forpin
> # - *981 - listen voice messages, being promted for mailbox andpin > # - *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); > > } > > >
> > _______________________________________________ > Users mailing list > Users@openser.org <mailto: Users@openser.org> > http://openser.org/cgi-bin/mailman/listinfo/users >-- Howard Tang ICQ : 259083 MSN : howard615@hotmail.com <mailto: howard615@hotmail.com>
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
-- Bill Neely Xantek, Inc. 1-866-553-3833 1-702-874-3833
-- Howard Tang ICQ : 259083 MSN : howard615@hotmail.com
-
Bill Neely -
Bogdan-Andrei Iancu -
Howard Tang