Hello everyone,
I am using the onsip call
features ser.cfg and am adapting it for asterisk voicemail. This is what I
currently have changed:
1)
In the usr_preferences table in the ser database have
an entry for
user 2092.
Insert into usr_preferences
(username, attribute, value) values
("2092", "voicemail",
"y");
2)
In Route[3] (used for call invite handling)
if(avp_db_load("$ruri/username","s:voicemail")){
if(avp_check("s:voicemail",
"eq/y/i")){
setflag(18);
};
};
This will check if the user wants to use
voicemail according to the
preference that
is set for them in the usr_preferences table. I they
don't want to
use voicemail set value to "n"
3) In failure route[1]
if (call fwd on
no answer is enabled{
} else if(isflagset(18) && t_check_status("408")){
route(x);
break;
};
4) route[x]
{
acc_db_request("missed
called", "missed_calls"); revert_uri();
rewritehostport("x.x.x.x:5064"); #port where asterisk is
listening
append_branch();
t_relay_to_udp(x.x.x.x",
"5064");
break();
}
I am getting a 404 sent back to the phone….I
suspect this is something got to do with route 1 as I have used loads of log
messages and I can see the flag being set, route x being called but after the
failure route, the code jumps to route 1…….This is probably because
in route 3 it says t_on_failure(“1”)
followed by route 4 followed by route 1…..I just don’t know what to
do about it…………Does anyone have any suggestions?
Kindest Regards,
Aisling.