17 May
2007
17 May
'07
8:44 a.m.
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> 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:
--
Howard Tang
ICQ : 259083
MSN : howard615(a)hotmail.com
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+Op…
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);
}
------------------------------------------------------------------------
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