[Users] Call forward when no answer
Bill Neely
ceo at xantek.cc
Thu May 17 21:16:47 CEST 2007
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
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