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:u...@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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