Hi,
I really do not know what is happening but when I'm sending SIP INVITE form
port other than 5060, if it reaches sl_send_reply(403,Forbidden)
response is sent to 5060 and not to the source port of the initial INVITE.
I am not changing anyhow $du or $rd ...
Mino
Via header
address.
Cheers,
Daniel
On 8/19/13 12:31 PM, Mino Haluz wrote:
Hi,
I really do not know what is happening but when I'm sending SIP INVITE
form port other than 5060, if it reaches sl_send_reply(403,Forbidden)
response is sent to 5060 and not to the source port of the initial
Ok guys, seems that no one interested in this, but I decided for http_query
with xmlops module - it gives quite good performance and reliability
(timeout, reply code checking).
On Mon, Jul 29, 2013 at 12:20 PM, Mino Haluz mino.ha...@gmail.com wrote:
Oh, I see, there is app_lua as well, so all
Hi,
we would like to remove any SQL in our configuration and to fetch all data
from web services (REST/RPC/whatever). What do you think, what would be the
best way to integrate webservices in kamailio? app_python? app_perl? When
calling any method from app_python configured script, it executes
Oh, I see, there is app_lua as well, so all I found that could be used for
calling webservices:
app_python
app_perl
app_lua
app_mono
http_query form utils module
So the question is, what would you use and why? :)
On Mon, Jul 29, 2013 at 10:09 AM, Mino Haluz mino.ha...@gmail.com wrote:
Hi
Hi,
I need to modify Allow header that is sent from Freeswitch, to restrict it
to only some very important ones. Do you think it really violates RFC ? :))
M
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Hi,
I've been playing with dispatcher settings, and I cannot get it working as
expected. I would like to have my destination IP1 being always probed and
some destinations should be marked as active non-probing.
So, in my config I have
ds_probing_mode = 0 (documentation: If set to 0, only the
to find some time for that NOPROBE thing, the
cases we need could be implemented, it would be awesome.
On Wed, Jun 19, 2013 at 2:44 PM, Daniel-Constantin Mierla mico...@gmail.com
wrote:
Hello,
On 6/19/13 1:57 PM, Mino Haluz wrote:
Hi,
I've been playing with dispatcher settings
http://asylum.madhouse-project.org/projects/syslog-ng/mongodb/
On Thu, Jun 20, 2013 at 12:25 AM, Daniel-Constantin Mierla
mico...@gmail.com wrote:
Hello,
Kamailio sends the logs to syslog, perhaps you can find something on the
net that allows syslog application to write to mongodb. I know
Hi,
I want to do this:
2 hosts, with 2 running kamailios, every host has 1 IP address and hostA
has virtualIP assigned. kamailio should run on both hosts. I made a script
which can transfer virtual IP from hostA to hostB.
The problem is, I cannot tell kamailio to use virtualIP on hostB because,
Mino
I have set up this scenario using heartbeat on both hosts and it works
wonderfully.
Regards
Phillip
Message: 1
Date: Fri, 14 Jun 2013 08:15:28 +0200
From: Mino Haluz mino.ha...@gmail.com
To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
Users
...@evaristesys.comwrote:
Hello Mino,
On 06/14/2013 04:53 AM, Mino Haluz wrote:
If I bind to non-existent IP address, than there is problem with
sending OPTIONS.
Perhaps this can help?
echo 1 /proc/sys/net/ipv4/ip_**nonlocal_bind
This will allow you to bind to IPs that don't
Hi,
I'm struggling to solve one problem with 503 reply code.
Kamailio receives 503 and changes it to 500, with error in log
WARNING: script writer didn't release transaction
I inspected incoming 503 (from Cisco gateway), everything seems to be ok
except for Via:
SIP/2.0/UDP
, the warning is usually for requests, but is harmless
anyhow.
Changing 503 in 500 is from RFC specs, but you can disable it:
http://kamailio.org/docs/modules/stable/modules/tm.html#remap_503_500
Cheers,
Daniel
On 6/14/13 5:42 PM, Mino Haluz wrote:
Hi,
I'm struggling to solve one problem
be inconsistent a while.
If rtimer runs a route, any other processing is stopped until this route
ends? If so, I do not need any lock.
On Thu, Jun 13, 2013 at 9:15 AM, Daniel-Constantin Mierla mico...@gmail.com
wrote:
Hello,
On 6/12/13 9:04 AM, Mino Haluz wrote:
Hi,
I need to lock
Hi,
what this function in fact does? It only compares host from dispatcher list
and $si ? or it compares with source port as well ?
I'm distributing traffic across multiple gateways with dispatcher module,
these gateways create another leg and send it back to proxy. I would like
to compare it
Hi,
I need to update some hash tables inside kamailio, but this update should
be triggered externally.
I know there is xmlrpc or xhttp, but I am just curious if there is some
easier way how to run this route. I have to run it instantly, so setting
some shv and run this route if new invite comes
Hi,
I need to lock the hashtable but this
lock($sht(a));
does not work. Any hints?
Mino
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Hi,
I need to define variables as string explicitly.
What I have in the code:
in prefix_list is 123,456,789
$var(matched_prefix) = $(var(prefix_list){s.select,$var(i),,});
then in MAIN route:
$var(matched_prefix) = route(INCOMING_AUTH);
and this check
if ($var(matched_prefix) != nullprefix)
Hi,
I have to iterate string like 1,2,3,4 in kamailio. How could you do this? I
know we have s.select,index,separator but I dont know how to go through all
elements. Thanks,
Mino
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Hi,
if I do save(location) when receiving REGISTER, what is the header which
indicates the subscriber for which it will be registered ?
$fu?$au?$tu?
Thanks,
Mino
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.
Cheers,
Marius
On Sun, Feb 24, 2013 at 6:19 PM, Javi Gallart
jgall...@systemonenoc.comwrote:
Hello
you may try to do the remove/add operations in each branch_route
Regards
Javi
-Original Message-
From: Mino Haluz mino.ha...@gmail.com
Sender: sr-users-boun...@lists.sip-router.org
Hi,
I have some nagios scripts binded to kamctl fifo commands, as well as some
web scripts that are directly calling kamctl fifo statements. I do not know
why, but it sometimes returns
** ERROR: Error opening Kamailio's FIFO /tmp/kamailio_fifo [1] = ** ERROR:
Make sure you have the line
block and reading it in failure route block
(use avps for that case).
$var(...) is faster to use and does not need locking at all. These are
usually referred as script variable, but this term can be confused with all
the config file variables.
Cheers,
Daniel
On 2/12/13 3:13 PM, Mino Haluz
Hi,
what is the difference between shared and script variable ? Thanks
Mino
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You have to figure it somehow with adjusting the python code of
media-dispatcher. We have developed special /etc/init.d/kamailio grace
command, which waits for the moment, when there are no calls, so it can be
restarted. It queries mediaproxy every second.
In future we will not use mediaproxy
cluster
Cheers,
Daniel
On 1/25/13 8:18 PM, Rumen Mihailov wrote:
Hello Mino,
Sorry for the offtopic, what HA database solution do you use ?
Regards,
Rumen
On 25 Jan 2013 16:22, Mino Haluz mino.ha...@gmail.com wrote:
Hi,
I made some DB test on our HA cluster, and I got that for 1
Hi,
I think that it is more than likely not implemented, but is there any
event-route that is triggered just before event_route[dialog:start] ? I
need to check some security flags before the dialog is created. But it is
too early too check them in relay route..
Mino
Hi,
I made some DB test on our HA cluster, and I got that for 1 process I have
1000 queries/sec and for 11 processes 3000 queries/sec. That's why I would
to ask if every kamailio process that is spawned has its own database
connection or queries are put in the queue and executed by some main
and how to check null value taken from database?
$dbr(ra=[0,0]) == ?
On Mon, Jan 14, 2013 at 1:44 PM, Alex Balashov abalas...@evaristesys.comwrote:
if(defined $var(x))
Or, if checking for empty value:
if(strempty($var(x))
Mino Haluz mino.ha...@gmail.com wrote:
Hi,
how should I
Hi,
I would like to set my custom different debug levels (with flag?)
externally with kamctl command. So I neednt restart kamailio if I want to
enable/disable debug.
Which module should I use in that case?
Thanks,
Mino
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Ok I see that it is possible with pv module and shv_set MI command. How can
I get value of this shared variable inside code?
On Fri, Dec 21, 2012 at 1:17 PM, Mino Haluz mino.ha...@gmail.com wrote:
Hi,
I would like to set my custom different debug levels (with flag?)
externally with kamctl
Ok I did it like this, every xlog I'm calling is in the form:
xlog(L_INFO,XLOG: $ci [ROUTENAME] debuginformation);
and also after every sql_query I have
xlog(L_INFO,XLOG: $ci [ROUTENAME] SQL: select * from );
and when SIP message is received/sent
xlog(L_INFO,XLOG: $ci [MAIN] $mb);
Hi,
I know how to use sl_send_reply, but I would like to add Reason: header
to the reply generated. Is there any command or variable for that?
Thank you,
Mino
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Hi,
what things should I set in order to set my custom RPID. What I am doing:
remove_hf(Remote-Party-ID);
append_rpid_hf(, ;party=calling;id-type=subscriber;screen=no);
but RPID is only removed, second command does not append any rpid. Thanks
Mino
/voip-eavesdropping-counter-measurements.html
Best regards.
2012/11/27 Mino Haluz mino.ha...@gmail.com
Hi,
maybe it is not that kamailio related question, but I dont know any
other place with such good voip professionals ;) I have kamailio and
mediaproxy. Clients are BudgetTone 200
Hi,
maybe it is not that kamailio related question, but I dont know any other
place with such good voip professionals ;) I have kamailio and mediaproxy.
Clients are BudgetTone 200 (Grandstream) and CSipSimple. I am forcing
clients to use SRTP but it does not support adding any certificate on both
Hi,
I'm using dialog module and I set some dlg_var for every dialog. Is it
possible to get this variable value with MI command somehow ? I know to get
list of dialogs using dlg_list only. Thanks!
Mino
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Hi,
we are thinking about switching from siptrace to sipcapture + HEP
encapsulation. Do you know some reasons why we should switch to
sipcapture or why we should not ? ;) We have multiple SIP proxies and
Cisco gateways. In earlier versions of kamailio was siptrace blocking,
that's why we started
Hi,
I am doing some performance tests and I cannot force SIP INVITEs to
use mediaproxy within LAN, without any NAT. I'm using
engage_media_proxy or rtpproxy_manage but phones are still exchanging
voice data between them, not through the proxy. Is there some trick
how to forcibly tell kamailio to
My fault, typoo in dispatcher's IP address.
On Tue, Oct 9, 2012 at 3:25 PM, Mino Haluz mino.ha...@gmail.com wrote:
Hi,
I am doing some performance tests and I cannot force SIP INVITEs to
use mediaproxy within LAN, without any NAT. I'm using
engage_media_proxy or rtpproxy_manage but phones
Hi,
mediaproxy has dispatcher and relay, so it load balances the traffic
automatically. How it is done with rtpproxy ?
Thanks,
Mino
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Hi,
I'm doing the performance test with kamailio + RTPProxy, but I would
like to get the real calls count that the rtpproxy is serving. I don't
want to use value that I get from sipp.
So is there any management tool for rtpproxy, or should I get it
somewhere in kamailio config ?
Thanks,
Mino
Ok, I'm tagging dialogs with dlg_manage(), but even if the call ends,
it still keeps info about this dialog in list kamctl fifo dlg_list.
Should I somehow close the dialog when the BYE transaction is ended ?
Peter: Thanks for the tip! Really interesting. But I do not
understand, why also this
I'm using rtpproxy_manage, so I assume unforce_rtp is not needed.
On Thu, Sep 13, 2012 at 4:10 PM, Peter Lemenkov lemen...@gmail.com wrote:
2012/9/13 Mino Haluz mino.ha...@gmail.com:
Peter: Thanks for the tip! Really interesting. But I do not
understand, why also this list contains the calls
in the refinement one might
expect from a proper keyboard.
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/
Mino Haluz mino.ha...@gmail.com wrote:
I'm using rtpproxy_manage, so I assume
Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/
Mino Haluz mino.ha...@gmail.com wrote:
The results:
- rtpproxy calls count 280
- sipp calls count 2000
- iptraf on proxy 4.8MB/s
- G711a codec
So if my calculations are right (16kB/s per stream * 280
that could cause this error ? Otherwise the call
is initiated ok, but I really dont understand what is so strange to
kamailio in this INVITE.
On Thu, Sep 13, 2012 at 6:37 PM, Peter Lemenkov lemen...@gmail.com wrote:
2012/9/13 Mino Haluz mino.ha...@gmail.com:
You mean on the proxy side? I'm
Hi,
one number is registered on 2 phones. Phone1 has Always redirect set
to another number. When incoming call is initiated, Phone2 is ringing
and Phone1 sends 302 to the proxy. However the proxy does not send 302
to the caller (for ex. GW), but it waits for timeout of the Phone2.
Then the proxy
Hi,
I know that kamailio is SIP proxy, but is there any way how to implement
kamailio as SBC like OpenSIPS with B2BUA module ? I tried OpenSIPS with
this module, but it does not work with mediaproxy module.
Mino
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We are troubleshooting this issue almost for 2 days and we did not find the
solution yet. The thing is, that it does not enter even the config file (we
have some debug messages at the start, that do not show). Otherwise we do
not have any problem with ACK, but only this particular ACK is not
Hm, the same settings works on testing 3.2.1. But this is not the solution
- I have to get it working on 3.2.0...
On Tue, Mar 6, 2012 at 5:16 PM, Mino Haluz mino.ha...@gmail.com wrote:
Hi,
I have siptrace working but when I set duplicate_uri, no packet is sent to
the destination
Hi,
I have siptrace working but when I set duplicate_uri, no packet is sent to
the destination. Was there some bug fix or anything I have to set further?
There is nothing in syslog, siptrace locally is stored fine.
modparam(siptrace, db_url, mysql://localhost/kamailio) #
Database URL
On Thu, Feb 16, 2012 at 10:42 AM, Sammy Govind govoi...@gmail.com wrote:
which version are you using, there is no such condition in this page or is
it?
http://kamailio.org/docs/modules/3.1.x/modules_k/dispatcher.html#id2821010
On Thu, Feb 16, 2012 at 2:31 PM, Mino Haluz mino.ha...@gmail.com
) yes it is expected behaviour, you just need to do record routing as
usual, nothing special.
Cheers,
Daniel
On 2/13/12 11:43 AM, Mino Haluz wrote:
Hi,
our customers are using mostly UDP but some of them want to use TCP. The
problem is, I get various TCP errors in kamailio log and I do
), what is the output of kamctl fifo ds_list?
Do you see anything else in /var/log/messages at this point?
Peter
On Tue, 2011-12-20 at 11:05 +0100, Mino Haluz wrote:
nightly build 3.2.1.
compiled on 05:08:17 Dec 19 2011 with gcc 4.4.5
On Tue, Dec 20, 2011 at 10:49 AM, Peter Dunkley
Hi,
how can I get the reason code from reply in failure_route ? $hdr(reason)
points to the INVITE request..
Thanks,
Mino
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I meant the reason specified in the reply as : Reason: Q.850;cause=17 ,
that is not the SIP response code in fact. T_reply_code relates to the SIP
response code.
On Fri, Dec 23, 2011 at 12:35 PM, Daniel-Constantin Mierla
mico...@gmail.com wrote:
Hello,
On 12/23/11 11:44 AM, Mino Haluz wrote
Ok, T_rpl(hdr(reason)) did the thing, however thanks!
On Fri, Dec 23, 2011 at 2:01 PM, Mino Haluz mino.ha...@gmail.com wrote:
I meant the reason specified in the reply as : Reason: Q.850;cause=17 ,
that is not the SIP response code in fact. T_reply_code relates to the SIP
response code
Hi,
I have two gateways pinged by kamailio. The both are AP (active/probing),
when I cut the one gateway off, it becomes IP(inactive/probing) but the
event_route is not fired up. Am I missing something ?
event_route[dispatcher:dst-down] {
xlog(L_ERR, Destination down: $rm $ru ($du)\n);
}
in Kamailio 3.2.0 and later.
Regards,
Peter
On Tue, 2011-12-20 at 10:06 +0100, Mino Haluz wrote:
Hi,
I have two gateways pinged by kamailio. The both are AP (active/probing),
when I cut the one gateway off, it becomes IP(inactive/probing) but the
event_route is not fired up. Am I missing
Hi,
please, could someone specify the conditions when the dispatcher module
sends the INFO pings to the gateways ?
modparam(dispatcher, flags, 2)
modparam(dispatcher, dstid_avp, $avp(s:test))
modparam(dispatcher, dst_avp, $avp(AVP_DST))
modparam(dispatcher, grp_avp, $avp(AVP_GRP))
attrs=
version: kamailio 3.2.0 (i386/linux)
On Mon, Dec 19, 2011 at 1:29 PM, Daniel-Constantin Mierla mico...@gmail.com
wrote:
Hello,
is this at least version 3.2.1 (or the latest branch 3.2)?
What do you get with: kamctl fifo ds_list?
Cheers,
Daniel
On 12/19/11 12:49 PM, Mino Haluz
Dec 19 2011 with gcc 4.4.5
On Mon, Dec 19, 2011 at 2:29 PM, Daniel-Constantin Mierla mico...@gmail.com
wrote:
Hello,
On 12/19/11 1:35 PM, Mino Haluz wrote:
This is in the log file:
ERROR: dispatcher [dispatcher.c:640]: failover functions used, but AVPs
paraamters required are NULL
Ok, my mistake, the modparams related to dispatcher module were in !endif
that was not processed. In future, I will do not edit the cfg file without
code highlighting ..arrg
On Mon, Dec 19, 2011 at 3:56 PM, Mino Haluz mino.ha...@gmail.com wrote:
Updated to 3.2.1 (debian package) and still
Hi,
I would like to use the dispatcher module with algorithm 10 - call load
balancing. But there are multiple things that are a bit unclear to me. I
have multiple gateways that can serve different maximum number of calls.
1) Can I somehow set the maximum calls count for each dispatcher gateway,
Hi,
# sercmd dispatcher.list
error: 500 - command dispatcher.list not found
The module is loaded, dispatcher.list file exists. Am I doing something
wrong ? Thanks.
Mino.
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*Im Auftrag von *Mino Haluz
*Gesendet:* Donnerstag, 15. Dezember 2011 14:31
*An:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
Users Mailing List
*Betreff:* [SR-Users] How to get dispatcher list with sercmd
Do I have to load some specific module ?
ERROR: connect_unix_sock: connect(/tmp/sercmd_ctl): No such file or
directory [2]
On Thu, Dec 15, 2011 at 3:55 PM, Daniel-Constantin Mierla mico...@gmail.com
wrote:
On 12/15/11 3:45 PM, Mino Haluz wrote:
kamctl dispatcher dump
did the thing
tm.t_uac_start
tm.t_uac_wait
On Thu, Dec 15, 2011 at 4:14 PM, Mino Haluz mino.ha...@gmail.com wrote:
Do I have to load some specific module ?
ERROR: connect_unix_sock: connect(/tmp/sercmd_ctl): No such file or
directory [2]
On Thu, Dec 15, 2011 at 3:55 PM, Daniel-Constantin Mierla
mico
it should be reliable..
On Mon, Nov 21, 2011 at 11:43 AM, Alex Balashov
abalas...@evaristesys.comwrote:
On 11/21/2011 05:42 AM, Mino Haluz wrote:
I was using cdrtool (prepaid table) and callcontrol to limit
concurrent calls. In fact this is only limiting the outbound calls,
but I would like
Hi,
the scenario is like this:
- user1 makes a call to user2
- proxy sends receives INVITE and sends it to user2
- user2 does not respond at all (firewall for example)
- user1 after 5 seconds hangs up the call with CANCEL
- proxy sends to user1 200 canceling
CANCEL is not forwarded to user2. Ok
Hi,
I was using ndb cluster as the dabatase backend for my kamailio setup. The
thing is, mysql ndb cluster is difficult to maintain for me and it crashed
multiple times (desynchronization, datamemory limit reached, and other
strange reasons) and it is not that well documented (the user-base is
Hi,
is there any setting which could allow me to set maximum memory per module ?
As I am testing the carrier route module, I've added for testing purposes
100 000 rules sofar. When I start the kamailio, it gives me :
Oct 10 14:46:46 kamrouter /usr/sbin/kamailio[2182]: ERROR: carrierroute
:
On 10/10/2011 03:51 PM, Mino Haluz wrote:
Hi,
is there any setting which could allow me to set maximum memory per module
? As I am testing the carrier route module, I've added for testing purposes
100 000 rules sofar. When I start the kamailio, it gives me :
Oct 10 14:46:46 kamrouter
in the callcontrol daemon? Is there
any timeout? If so, it can be clearly fixed by this way ;)
Hope this helps to somebody,
Cheers
On Mon, Aug 15, 2011 at 9:59 AM, Mino Haluz mino.ha...@gmail.com wrote:
Any updates on this? I updated callcontrol which has some bug fixed:
callcontrol (2.0.14) unstable
Hi,
I have this code:
if ( is_user_in(From, blocked) is_method(INVITE)) {
xlog(L_INFO, XLOG: [number_and_ruri_checks] NOTICE:
Account ($fu) to ($ru) is blocked);
sl_send_reply(403, Account blocked );
exit;
}
I would like to
there is no record in db, it cannot be updated.
On Mon, Sep 26, 2011 at 12:15 PM, Daniel-Constantin Mierla
mico...@gmail.com wrote:
Hello,
On 9/26/11 11:26 AM, Mino Haluz wrote:
Hi,
I have this code:
if ( is_user_in(From, blocked) is_method(INVITE)) {
xlog(L_INFO, XLOG
Sorry for misunderstanding, yes, the failed status is firing the
insert_radacct_record but with different parameters. I must see why it is
not written to db.
Thanks.
On Mon, Sep 26, 2011 at 1:05 PM, Mino Haluz mino.ha...@gmail.com wrote:
Ok, but when I use acc_rad_request only Failed status
Hi,
I need a module which could allow me to send traffic to various carriers and
it has to support some important features. So some basic ones:
- possibility to re-route the call in case the original route fails
- peak/offpeak conditions (time-based)
- route traffic according to prefix
I found
Hi,
I would like to resend the INVITE to another gateway if the original
fails (503, 500, etc.). Is there any module or mechanism which could
do this ?
Mino
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(see the
config below).
t_newtran before calling callcontrol does not work as well - it is
executed 3 times.
On Mon, Jun 13, 2011 at 4:29 PM, Mino Haluz mino.ha...@gmail.com wrote:
It does not work, t_newtran always returns success, so it will never
absorb the retransmission.
So what I did
Hi,
does kamailio support conference calls or it has to be implemented in
some B2BUA ?
If it is supported, what are the requirements (modules etc.).
Thanks,
Mino
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handling it to call control to be sure the retransmitted requests are
absorbed.
http://kamailio.org/docs/modules/stable/modules/tm.html#t_newtran
Cheers,
Daniel
On 6/6/11 4:29 PM, Mino Haluz wrote:
Hi,
my kamailio server is receiving from some customers 3 identical INVITEs when
call
callcontrol.
That is why, the prepaid limit is not working at all in this case. This way
the user can hack the prepaid protection of the account. Otherwise the
call_control is fuilly functional.
Anybody experienced the similar problem? If so, how to resolve it?
Thanks,
Mino Haluz
Hi,
I see in my cdrtool rarely 422 Session Timer too small - the reason why the
call was not initiated. Do you have some suggestion how to fix this ? Maybe
SST module ? What is the default value of this timer, so that I could
increase it ? kamailio 3.1.0 (i386/linux) 1e204f
Thank you,
Mino
Hi,
I have a question about kamailio-asterik interconnection. I'd like to
connect 1000 numbers with a trunk, but it would be painful to add 1000
trunks on asterisk. Do you have some idea how could I group those numbers
into one trunk connected to kamailio? asterisk and kamailio configuration
if this way is reported only once.
Be sure flag 4 is not used for something else, of if it used then use an
unused one.
Cheers,
Daniel
On 1/18/11 11:41 AM, Mino Haluz wrote:
These are my modparams.
modparam(acc, failed_transaction_flag, 1)
modparam(acc, report_cancels, 1)
modparam
Hi,
I would like to force kamailio to send another code as Request timeout when
fr invite timeout is hit. Is there some nice way how to achieve it, or I
have to edit the code ? :(
Thank you!
___
SIP Express Router (SER) and Kamailio (OpenSER) -
Hi,
I'm using radius server to do an accounting, but it does not store an
application type (audo/video). Audio/video calls has to be rated with
cdrtool, that is why I need this information. It is not even sent through
radius protocol, so the radius server cannot get it. My extra-params are:
These are some lines from syslog. Where could I find the problem ?
Oct 12 15:13:15 nod2 /usr/local/sbin/kamailio[23102]: DEBUG: core
[sr_module.c:625]: find_export_record: found load_tm in module tm
[/usr/local/lib/kamailio/modules/tm.so]
Oct 12 15:13:15 nod2 /usr/local/sbin/kamailio[23102]:
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