[Users] Call forward when no answer
Howard Tang
howard615 at gmail.com
Fri May 18 01:37:14 CEST 2007
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 at 10.10.1.2 F=
sip:0299000000 at 10.10.1.2 T=sip:0280000000 at 10.10.1.2 IP=10.10.1.1 ID=
call-F11EC874-4CE7-2910-000A-3E6 at 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("$ruri", "i:10");
prefix("1");
# route to Asterisk Media Server
rewritehostport("10.10.10.11:5060");
resetflag(2);
route(1);
}
May 18 09:08:45 localhost /usr/sbin/openser[24414]: failure_route - call
forward to Voice Mail - M=INVITE
RURI=sip:0280000000 at 10.10.2.126:57042;rinstance=dbdab29df7aa260b
F=sip:0299000000 at 10.10.1.2 T=sip:0280000000 at 10.10.1.2 IP=10.10.1.1 ID=
call-F17BFBB3-4FE7-2910-000C-3E8 at 10.10.1.1
May 18 09:08:59 localhost /usr/sbin/openser[24399]:
ERROR:tm:t_forward_nonack: no branch for forwarding
May 18 09:08:59 localhost /usr/sbin/openser[24399]: ERROR:tm:w_t_relay:
t_forward_nonack failed
May 18 09:09:09 localhost /usr/sbin/openser[24399]:
ERROR:tm:t_forward_nonack: failure to add branches
Anyone have an idea on where i have done wrong?
Regards,
Howard
On 5/18/07, Bill Neely <ceo at xantek.cc> wrote:
>
> I am having a very similar problem. Using v1.2.0
>
> Here is my route:
> route[1] {
>
> if(isflagset(2))
> t_on_failure("2");
>
>
> if (!t_relay()) {
> sl_reply_error();
> };
> exit;
> }
>
> failure_route[2]
> {
> if ( t_check_status("408"))
>
> {
> xlog("L_ERR","rrreeeeeeeeeeeeeeeeeecalling froute2 <$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><$ruri>\n");
> route(1);
>
>
> }
> exit;
> }
>
> Here is error message received:
>
> 1(53165) rrreeeeeeeeeeeeeeeeeecalling froute2
> <INVITE><sip:1020101 at 67.188.xxx.188:35937;rinstance=e867c589f1896b12>
> 1(53165) 22222222222222222222calling froute2
> <INVITE><sip:7771020101 at 66.xxx.20.50:5060;rinstance=e867c589f1896b12>
> 1(53165) ERROR:tm:t_forward_nonack: no branch for forwarding
> 1(53165) ERROR:tm:w_t_relay: t_forward_nonack failed
>
> Bogdan-Andrei Iancu wrote:
> > Check with log/xlog prints if it gets to t_on_failure() and into
> > failure route.
> >
> > regards,
> > Bogdan
> >
> > Howard Tang wrote:
> >> HI Bogdan,
> >>
> >> Thank you for your reply. I did that but i forget to include in this
> >> email.
> >>
> >>
> >> route[1] {
> >> #check for nat flag
> >> if (isflagset(2))
> >> {
> >> fix_nated_contact();
> >> use_media_proxy();
> >> }
> >>
> >> t_on_reply("1");
> >> t_on_failure("1");
> >>
> >> # send it out now; use stateful forwarding as it works reliably
> >> # even for UDP2TCP
> >> xlog("L_INFO", "Request leaving server - M=$rm RURI=$ru F=$fu
> >> T=$tu IP=$si ID=$ci\n");
> >> if (!t_relay()) {
> >> if(isflagset(2))
> >> end_media_session();
> >> sl_reply_error();
> >> };
> >> exit;
> >> }
> >>
> >> The voice mail work fine only when someone call in and the UA is
> >> offline (not registered to the openser), if the UA is online, the
> >> call will ring the UA until the caller hang up.
> >>
> >> I want to set up some sort of timer, i.e. 60 second and the call will
> >> forwarded to the Voice mail.
> >>
> >> Can you suggest me an idea on how i can make this happen please?
> >>
> >> Regards,
> >> Howard
> >>
> >>
> >>
> >> On 5/17/07, *Bogdan-Andrei Iancu* <bogdan at voice-system.ro
> >> <mailto:bogdan at voice-system.ro>> wrote:
> >>
> >> Hi Howard,
> >>
> >> I guess you do not arm the failure route - use t_on_failure("1");
> >> before
> >> relaying the request.
> >>
> >> regards,
> >> bogdan
> >>
> >> Howard Tang wrote:
> >> > Hi All,
> >> >
> >> > I have followed a tutorial and set up Asterisk as a voice mail
> >> server.
> >> >
> >> >
> >>
> >>
> http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+With+OpenSER
> >>
> >> >
> >>
> >> <
> http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+With+OpenSER
> >>
> >>
> >> <
> http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+With+OpenSER
> >>
> >>
> >> >
> >> > It works fine when the UA is offline. Now, I want a call
> >> forwarded to
> >> > the Voice mail server when there is no answer from the UA after
> 60
> >> > seconds(UA is registered on the openser).
> >> >
> >> > What should I do? Below is my config (copy from the above link).
> >> >
> >> >
> >> > # requests for Media server
> >> > if(is_method("INVITE") && !has_totag() &&
> >> uri=~"sip:\*9") {
> >> > route(3);
> >> > exit;
> >> > }
> >> >
> >> > # mark transaction if user is in voicemail group
> >> >
> >> > if(is_method("INVITE") && !has_totag()
> >> > && is_user_in("Request-URI","voicemail"))
> >> > {
> >> > xdbg("user [$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 at openser.org <mailto:Users at openser.org>
> >> > http://openser.org/cgi-bin/mailman/listinfo/users
> >> >
> >>
> >>
> >>
> >>
> >>
> >>
> >> --
> >> Howard Tang
> >> ICQ : 259083
> >> MSN : howard615 at hotmail.com <mailto:howard615 at hotmail.com>
> >
> >
> > _______________________________________________
> > Users mailing list
> > Users at 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 at hotmail.com
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