[Kamailio-Users] Problems in ACC with missing BYE

[David]

Hey,

Thanks for the answer. If I did not have Kamailio, how would I do this?

David

Uriel Rozenbaum wrote:
> Hi David,
>
> Maybe you can set rtptimeout on Asterisk peer, so when no RTP is 
> flowing, Asterisk will hang up the call and you'll have the CDR 
> "closed" in Kamailio.
>
> Be sure your Kamailio is redundant, you can use heartbeat or something 
> like that.
>
> Rgds,
> Uriel
>
> On Thu, Jun 11, 2009 at 10:08 AM, David <kamailio.org 
> <http://kamailio.org>@spam.lublink.net <http://spam.lublink.net>> wrote:
>
>     Hi,
>
>     I am using Kamailio as my ACC, Dispatcher, far end nat and
>     presence server in front of a farm of asterisk boxes.
>
>     Most calls are being properly added into my acc table and using a
>     join between the INVITEs, CANCELs, and BYEs I am able to get what
>     seems like accurate call detail records.
>
>     The trouble is that every so often a BYE does not make it back to
>     my server. In my simulation this morning, I simply unplugged (
>     electric ) the two phones that were having a pleasant
>     conversation.  Now I have asterisk that thinks the call is still
>     running and I have Kamailio which has no ending 'BYE' message. For
>     the most part this is not a big deal, but when I can a cellular
>     phone in European countries, my provider thinks I am still
>     talking. At 30 cents a minute, that's a lot.
>
>     Here are some snippets from my code :
>
>     loadmodule("dialog.so")
>     loadmodule("acc.so")
>     loadmodule("sst.so")
>
>     modparam("acc", "early_media", 1)
>     modparam("acc", "report_ack", 1)
>     modparam("acc", "report_cancels", 1)
>     modparam("acc", "failed_transaction_flag", 3)
>     modparam("acc", "log_flag", 1)
>     modparam("acc", "log_missed_flag", 2)
>     modparam("acc", "db_flag", 1)
>     modparam("acc", "db_missed_flag", 2)
>     # There is also a parameter for the DB, but I can't give you my
>     password
>     modparam("acc", "db_url", "some://valid:url@to/db")
>
>     # Note $avp(i:10) always ends up being 14400 ( less than the value
>     on the help page )
>     modparam("dialog", "timeout_avp", "$avp(i:10)")
>     modparam("sst", "timeout_avp", "$avp(i:10)")
>     modparam("sst", "sst_flag", 5)
>
>
>
>     Relevant snippets from my routing :
>
>     if ( has_totag()) {
>     if ( loose_route() ) {
>      if ( is_method("CANCEL|BYE") {
>       setflag(1);
>       setflag(3);
>
>     }
>     }
>
>     # Routing of INVITEs
>     setflag(2)
>     if ( !is_method("ACK"))
>     {
>      setflag(1);
>     }
>
>
>
>     setflag(4);
>
>     setflag(5);
>
>
>     For invites, I have a onreply_route and failure_route which I use
>      only for RTP Stuff.
>
>     On reply route checks if rtpproxy is needed, if it is it is
>     activated. failure_route checks if rtpproxy was activated and if
>     it was deactives it. The only other code in the failure route is
>     this :
>
>     if ( t_was_cancelled() ){
>     exit ;
>     }
>
>     So, the problem is, when phones do not send BYE, what do I do? I
>     need resources freed up from Asterisk, RTP Proxy, and Kamailio
>     Dialog, and I need the call to be canceled with my provider and I
>     need for my ACC to recieve some indication as to when the call
>     ended. Obviously it won't be exact to the second, but I kind of
>     thought that the SIP Session Timers would notice the phone was
>     gone and would generate a BYE or something?
>
>     What do I do?
>
>     Thanks,
>
>     David
>
>
>
>
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