[Serusers] voicemail route

Greger V. Teigre greger at teigre.com
Tue Oct 4 07:54:51 CEST 2005


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 
  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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