[Kamailio-Users] Problems in ACC with missing BYE
Uriel Rozenbaum
uriel.rozenbaum at gmail.com
Thu Jun 11 18:10:40 CEST 2009
I'm hot sure I understood your last question. The trick is to let some
RTP-aware device on the flow hangup the call. In the general section of your
sip.conf (asterisk) you can config rtptimeout and rtpholdtimeout parameters
to 30 seconds; if no RTP is coming for that period of time, the call will be
dropped.
On Thu, Jun 11, 2009 at 1:05 PM, David <kamailio.org at spam.lublink.net>wrote:
> 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
>>
>>
>>
>>
>> _______________________________________________
>> Kamailio (OpenSER) - Users mailing list
>> Users at lists.kamailio.org <mailto:Users at lists.kamailio.org>
>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>
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