12 Feb
2013
12 Feb
'13
11:30 a.m.
---------- Forwarded message ----------
From: Bayram Karagoz <karagoz.bayram(a)gmail.com>
Date: 2013/2/12
Subject: Re: [SR-Users] INVITE Retransmission Causes Limiting Simultaneus
Calls for Subscribers
To: miconda(a)gmail.com
Cc: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users
Mailing List" <sr-users(a)lists.sip-router.org>
recheck config and saw that I used dlg_manage() most before route(QUOTA)
block. does it sits in the wrong place?
you can find my kamailio.cfg file in attachment also.
# account only INVITEs
* dlg_manage();*
if (is_method("INVITE"))
{
setflag(FLT_ACC); # do accounting
}
###Display Name manipulation###
if (is_method("INVITE") && ($fn != $fU)) {
uac_replace_from("$fU","");
}
### incoming call from SBC###
route(INCOMINGSBC);
###Suspend User###
route(SUSPEND);
### Concurrent Call Limitation Routing Block ###
* route(QUOTA);*
2013/2/12 Daniel-Constantin Mierla <miconda(a)gmail.com>
Where do you use dlg_manage()? It is not in the route
you pasted below.
Cheers,
Daniel
On 2/12/13 10:12 AM, Bayram Karagoz wrote:
Thanks for quick response Daniel,
My kamailio version is 3.2.3 (x86_64/linux) .
I want to use dlg_manage() before but not sure which side of my call
limitation route code block.
can you advise me where to use dlg_manage() in my routing?
here is my route block defined below;
route[QUOTA] {
if (is_method("INVITE") && $fU == $avp(s:username)) {
avp_db_query("select quota from subscriber where
username='$avp(s:username)'", "$avp(s:quota)");
$var(SIZE) = 0;
get_profile_size("callquota", "$avp(s:username)",
"$var(SIZE)");
xlog("L_INFO", "CALL_KAMAILIO:Call originated from =
$fd\n");
xlog("L_INFO", "Active call amount for $fU is $var(SIZE)
,
call-id=$ci, quota=$avp(s:quota)\n");
if( $var(SIZE) >= $avp(s:quota) ){
sl_send_reply("403", "Simultaneous calls limit
reached");
xlog("L_INFO", "QUOTA:Limit of $var(SIZE) active calls has
been reached for user=$avp(s:username), call-id=$ci,
quota=$avp(s:quota)\n");
exit;
}
set_dlg_profile("callquota","$avp(s:username)");
}
}
2013/2/12 Daniel-Constantin Mierla <miconda(a)gmail.com>
--
*Bayram KARAGÖZ*
*NGN Engineer*
*-----------------------------------------*
*@* bayram(a)bayramkaragoz.org
*:** *www.bayramkaragoz.org
*-----------------------------------------*
--
*Bayram KARAGÖZ*
*NGN Engineer*
*-----------------------------------------*
*@* bayram(a)bayramkaragoz.org
*:** *www.bayramkaragoz.org
*-----------------------------------------*
Hello,
what version of kamailio are you using?
State 1 is for dialogs that were not relayed (or they haven't received
any reply yet). That means you create the dialog and don't relay the
INVITE. Try to us dlg_manage() just before relaying the INVITE.
Cheers,
Daniel
On 2/12/13 9:37 AM, Bayram Karagoz wrote:
Hi,
is there any advise about my problem?
I'm using kamailio in my production environment and this issue started to
reproduce a lot.
is there any way to discard retransmitted INVITE messages from counting
dialog?
2013/2/9 Bayram Karagoz <karagoz.bayram(a)gmail.com>
--
*Bayram KARAGÖZ*
*NGN Engineer*
*-----------------------------------------*
*@* bayram(a)bayramkaragoz.org
*:** *www.bayramkaragoz.org
*-----------------------------------------*
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
Kamailio World Conference, April 16-17, 2013, Berlin
- http://conference.kamailio.com -
thanks for the response Daniel,
has_totag not solved my problem.
in a deep search I realized that these retransmission INVITE dialogs
are waiting in dialog list as state = 1. one example is below;
is there any way to clear these dialog which state is waiting for state
= 1 or any other solution is welcome for me?
dialog:: hash=246:499719317
state:: 1
ref_count:: 1
timestart:: 0
timeout:: 0
callid:: 1356619-3569384775-474862(a)GBSBC-1.mydomain.com
from_uri:: sip:02165151057@xxx.xxx.xxx.xxx
from_tag:: 3569384775-474868
caller_contact:: sip:02165151057@xxx.xxx.xxx.xxx:5060
caller_cseq:: 2
caller_route_set::
caller_bind_addr:: udp:xxx.xxx.xxx.xxx:5060
callee_bind_addr::
to_uri:: sip:4115667@xxx.xxx.xxx.xxx
to_tag::
callee_contact::
callee_cseq::
callee_route_set::
2013/2/4 Daniel Tryba <daniel(a)pocos.nl>
--
*Bayram KARAGÖZ*
*NGN Engineer*
*-----------------------------------------*
*@* bayram(a)bayramkaragoz.org
*:** *www.bayramkaragoz.org
*-----------------------------------------*
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
Kamailio World Conference, April 16-17, 2013, Berlin
- http://conference.kamailio.com -
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
On Sunday 03 February 2013 12:58:02 Bayram
Karagoz wrote:
> I'm trying to limit all subscriber's active call according to setting
their
> simultaneous use setting defined in database.
> But having problem when one subscriber's INVITE message goes to
> retransmission. When retranmission occures INVITE messages don't
complete
> with BYE and subscriber simultaneus use amount will hung.
> My example code block is below.
> Does anyone advice me for overcome INVITE retransmissions?
I haven't implemented it myself yet, but all examples I've seen check
for
!has_totag() in the INVITE when doing call limit accounting. So
if (is_method("INVITE") && $fU == $avp(s:username) &&
!has_totag())
in your case. But that only works if the device received the initial 100
Trying with a tag from Kamailio.
But might it not be better to just do the count in the INVITE and
adding/deleting of dialogs in the dialog event routes?
http://kamailio.org/docs/modules/stable/modules_k/dialog.html#id2556156
Problem here is that if the limit is 5 and and there are more than 5
concurrent INVITES you'd have to drop calls with a 200 OK response.
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--
*Bayram KARAGÖZ*
*NGN Engineer*
*-----------------------------------------*
*@* bayram(a)bayramkaragoz.org
*:** *www.bayramkaragoz.org
*-----------------------------------------*