17 May
2007
17 May
'07
9:18 p.m.
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("$ruri", "i:10");
prefix("1");
# route to Asterisk Media Server
rewritehostport("
10.10.10.11:5060");
resetflag(2);
append_branch();
route(1);
}
Regards,
Howard
On 5/18/07, Howard Tang <howard615(a)gmail.com> wrote:
I have the same problem. It works fine if the forward happened in
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
if(isflagset(2)) {
xlog("L_INFO", "Callee is Offline, call
forward to Voice Mail - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
# route to Asterisk Media Server
prefix("1");
rewritehostport("10.10.10.11:5060
");
route(1);
} else {
sl_send_reply("404", "Not Found");
exit;
}
May 18 09:06:21 localhost /usr/sbin/openser[24410]: Callee is Offline,
call forward to Voice Mail - M=INVITE RURI= sip:0280000000@10.10.1.2 F=
sip:0299000000@10.10.1.2 T=sip:0280000000@10.10.1.2 IP= 10.10.1.1 ID=
call-F11EC874-4CE7-2910-000A-3E6(a)10.10.1.1
It is not working good in Failure_route
failure_route[1] {
if (t_was_cancelled()) {
xdbg("transaction was cancelled by UAC\n");
return;
}
xlog("L_INFO", "failure_route - call forward to Voice Mail - M=$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(a)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(a)xantek.cc> wrote:
--
Howard Tang
ICQ : 259083
MSN : howard615(a)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:
--
Bill Neely
Xantek, Inc.
1-866-553-3833
1-702-874-3833
--
Howard Tang
ICQ : 259083
MSN : howard615(a)hotmail.com
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(a)voice-system.ro
> <mailto: bogdan(a)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+Op…
>
> >
>
> <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);
> >
> > }
> >
> >
> >
>
>
------------------------------------------------------------------------
>
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>
--
Howard Tang
ICQ : 259083
MSN : howard615(a)hotmail.com <mailto: howard615(a)hotmail.com>
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