11 Jun
2009
11 Jun
'09
9:58 a.m.
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(a)spam.lublink.net>wrote;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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