[Serusers] voicemail route

[Aisling]

Ok Greger, thanks for the reply. 
 
Just a quick question, for the forward to voicemail on busy scenario, I
can see the callee phone sending the 486 busy message to SER and SER
sends an ACK back. At this stage I then see SER send a 404 to the
caller.I presume the correct sequence instead for this would be SER
forwarding the INVITE to Asterisk?
 
Many thanks.
 
-----Original Message-----
From: Greger V. Teigre [mailto:greger at teigre.com] 
Sent: 04 October 2005 06:55
To: Aisling; serusers at lists.iptel.org
Subject: Re: [Serusers] voicemail route
 
Aisling,
I think the only way you can get further on this is to use ngrep and
create a complete trace of the call. Then you have to match each of your
log messages to each SIP message. sip_scenario can help you in drawing
out who sent what.  Remember that once you relay to Asterisk, Asterisk
will get in the loop and these messages should also be relayed properly.
My guess is that this has something to do with the OK or ACK at the end
of call. Most likely you forget about a SIP message when reading your
logs... ;-) (I've done it myself so many times)
g-)
----- Original Message ----- 
From: Aisling <mailto:ashling.odriscoll at cit.ie>  
To: serusers at lists.iptel.org 
Sent: Monday, October 03, 2005 09:05 PM
Subject: [Serusers] voicemail route
 
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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