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.
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