[Serusers] voicemail route

Iqbal iqbal at gigo.co.uk
Tue Oct 4 13:01:38 CEST 2005


if ser is sending you the 404, then what is asterisk sending ser

what I normally do is sip debug on asterisk and print it out, then ngrep 
on ser also...or just ngrep both paths, and draw it out sip_scenario is 
good, but a bit OTT sometimes

Iqbal

Greger V. Teigre wrote:

> 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 <mailto: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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