i have been using radius for accounting, but now looked at db accounting
for another project. couple of issues:
- why is sip reason (the comment after reply's reason code) a mandatory
field in accounting table? if someone offers a good reason, my
opinion is that it should be removed. if someone needs it, he/she can
add it to db_extra list.
- is there possibility to fall back to syslog accounting if database
insert fails?
-- juha
Hi!
I wonder what is the status of local routes (to modify requests created
by Kamailio)? Is it implemented at all?
I found just one occurrence of USE_LOCAL_ROUTE in the code. (something
obsolete)?
thanks
Klaus
Hey Guys,
I came in need to delay 200 OK sent to originator, I know that is not
the job of a proxy (not a perfect world though), so if anyone think
that should be in theory at least possible with kamailio/sr, I would
appreciate any tip.
Scenario: User A -> Proxy -> User B.
User B will send 183 and then 200 to the User A.
I would need to delay the 200 OK sent to User A with a number of
seconds, but still send ACK to User B to "fake" the receiving of the
message in time. When the ACK for the delayed 200 OK comes back from
User A, I would simply absorb it.
Would that be possible with kamailio/sr without using an external application?
Ta for any kind of tip!
DanB
rls is extra_excluded in debian rules file:
# extra modules to skip, because they are not compilable now
# - regardless if they go to the main kamailio package or to some module
package,
# they will be excluded from compile and install of all
EXTRA_EXCLUDED_MODULES=bdb dbtext oracle pa rls iptrtpproxy
I tested and modules_k/rls compiles fine. I think the above text is true
only for ser's version.
regards
Klaus
I am using the default script;
When I do a lookup for a user that is not registered; I get 404 back;
I added in my failure route to the list of codes 404; ie:
First on INVITE I have
t_on_failure("FAIL_ONE");
failure_route[FAIL_ONE] {
if (t_check_status("486|408|404")) {
sethostport("mydomain.com:5060");
append_branch();
}
}
But for some reason I dont see it getting caught in the failure route
and sending to the server specified;
Thanks.
I have also tried the same using the following: one Kamailio server box and
one Asterisk server box. Same results after following the tutorial, with
exception of using two boxes.
Kurt A. Mullen
Practical PC, LLC
(O) 830.542.4102 x204
(F) 210.767.3912
I have two separate servers. A Kamailio box and an Elastix box.
I have Kamailio all up and running. I installed Elastix 2.0 using an ISO.
I followed all the steps to modify the Asterisk configuration from Kamailio
3.1.x and Asterisk 1.6.2 Realtime Integration using Asterisk Database to
provide access to the asterisk database.
1) I am not able to register an X-lite softphone to Kamailio.
a. Is this because Kamailio is not talking to the Asterisk DB or,
would X-Lite connect to Kamailio anyway?
b. kamailio.cfg :
i.
#!ifdef WITH_ASTERISK
ii.
asterisk.bindip = "192.168.15.211" desc "Asterisk IP Address"
iii.
asterisk.bindport = "5080" desc "Asterisk Port"
iv.
kamailio.bindip = "192.168.15.249" desc "Kamailio IP Address"
v.
kamailio.bindport = "5060" desc "Kamailio Port"
vi. #!endif
2) From the Kamailio server I am able to make an MYSQL connection to
the remote Elastix server.
a. #!define DBURL "mysql://openser:openserrw@localhost/openser"
b. #!define DBASTURL
"mysql://asterisk:asterisk1017@'192.168.15.211'/asterisk"
3) From the Elastix server I am able to make a MYSQL connection to the
remote Kamailio server (not sure this is necessary)
Some items are unclear to me in the tutorial.
1) The first two steps, MYSQL INSTALLATION and INSTALL UnixODBC is
this just on the Kamailio server, the Asterisk Server or both.
2) After creating the database on the Kamailio server
"/usr/local/sbin/kamdbctl create" is the asterisk database configured on the
Asterisk server?
3) After creation of the asterisk database, is the ODBC configuration
on the asterisk server, Kamailio server, or both?
4) Note: "Be sure you configure Asterisk to not authenticate SIP
requests coming from Kamailio". How do you do this?
Kurt A. Mullen
Practical PC, LLC
(O) 830.542.4102 x204
(F) 210.767.3912
-----Original Message-----
From: sr-users-bounces(a)lists.sip-router.org
[mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of
sr-users-request(a)lists.sip-router.org
Sent: Monday, December 13, 2010 5:39 AM
To: sr-users(a)lists.sip-router.org
Subject: sr-users Digest, Vol 67, Issue 31
Send sr-users mailing list submissions to
sr-users(a)lists.sip-router.org
On 12/9/10 7:20 PM, Kurt Mullen wrote:
> This issue has not been resolved. I am still getting the same error when
> trying to use kamctl for anything.
>
> I have tried all of the suggestions posted to date with no resolution. Is
> there anything more I can provide to help resolve this problem?
can you post here the parameters of mi_fifo module in kamailio.cfg,
content of kamctlrc related to fifo and output of 'ls -la /tmp | grep
kamailio'?
Cheers,
Daniel
--
Daniel-Constantin Mierla
Kamailio (OpenSER) Advanced Training
Jan 24-26, 2011, Irvine, CA, USA
http://www.asipto.com
[[Kurt Mullen]] kamailio.cfg
[[Kurt Mullen]] loadmodule "mi_fifo.so"
# ----- mi_fifo params -----
modparam("mi_fifo", "fifo_name","/tmp/kamailio_fifo")
modparam("mi_fifo", "fifo_mode",0666)
kamctlrc
CTLENGINE="FIFO"
## path to FIFO file
OSER_FIFO="/tmp/kamailio_fifo"
root@Kamailio:/# 'ls -la /tmp | grep kamailio'
-bash: ls -la /tmp | grep kamailio: No such file or directory
------------------------------
Hello,
where can I change/configure the location of file "kamailio.cfg"?
Reason: We like to use different "kamailio.cfg" files depending on the user that starts the kamailio.
Thank you...
Regards,
Nicolas
--
Neu: GMX De-Mail - Einfach wie E-Mail, sicher wie ein Brief!
Jetzt De-Mail-Adresse reservieren: http://portal.gmx.net/de/go/demail
remove
--- On Mon, 12/13/10, sr-users-request(a)lists.sip-router.org <sr-users-request(a)lists.sip-router.org> wrote:
From: sr-users-request(a)lists.sip-router.org <sr-users-request(a)lists.sip-router.org>
Subject: sr-users Digest, Vol 67, Issue 33
To: sr-users(a)lists.sip-router.org
Date: Monday, December 13, 2010, 6:48 AM
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When replying, please edit your Subject line so it is more specific
than "Re: Contents of sr-users digest..."
Today's Topics:
1. Re: Dialog module with 2 servers and 2 separate databases.
(Daniel-Constantin Mierla)
2. First public release of sip:provider Community Edition
(Andreas Granig)
3. First public release of sip:provider Community Edition
(Andreas Granig)
4. Re: Crash (michel freiha)
----------------------------------------------------------------------
Message: 1
Date: Mon, 13 Dec 2010 15:12:13 +0100
From: Daniel-Constantin Mierla <miconda(a)gmail.com>
Subject: Re: [SR-Users] Dialog module with 2 servers and 2 separate
databases.
To: "Pan B. Christensen" <pan(a)ibidium.no>
Cc: sr-users(a)lists.sip-router.org
Message-ID: <4D0629BD.8050501(a)gmail.com>
Content-Type: text/plain; charset="iso-8859-1"; Format="flowed"
Hello,
one option you can do is to write to db immediately when a call is
active (see dialog module parameters) and do a query to the other server
database in addition to counting the local instance active calls.
Another one, different, is to use memcache for a communication system
between two or more instances.
Cheers,
Daniel
On 12/13/10 2:37 PM, Pan B. Christensen wrote:
> Thanks for your reply, Daniel.
> The purpose is to do busy forwarding without querying the client when
> the user already has >= X active calls. X will normally be 1 (call
> waiting inactive) or 2 (call waiting active). Advanced users may
> possibly set a higher value. Counting the number of calls on
> the server and doing busy forwarding based on that rather than waiting
> for a "486 Busy here" from the client has several advantages.
> Currently, I've written code to do this with the dispatcher module,
> and it's working great with only one server. Here's a code snippet:
> $var(dlg_busy) = 0;
> get_profile_size("busy", "$avp(s:uid)", "$var(dlg_busy)");
> if ( $var(dlg_busy) >= $avp(s:busy_level) ) {
> if ($avp(s:cfb_status) == "on") {
> $rU = $avp(s:cfb_number);
> xlog("L_INFO", "-------------------- $avp(s:uid) has
> $var(dlg_busy) active calls. Treshold $avp(s:busy_level). Forwarding
> on busy to $rU --------------------\n");
> route(10);
> }
> ...
> }
> Based on your reply, I guess one way to solve this would be to write
> the get_profile_size function in sqlops, query the two dialog
> databases and add the numbers. This would still require the customer
> to change their database design. Is there an easier or better way to
> do this?
> I also wote code to do busy forwarding if the client replies with 486
> (do not disturb activated), 603 (call rejected) etc.
> This code works for normal busy forwarding if Polycom is set to 1 call
> per line key (default is 8). We'll then have to provision the
> $avp(s:busy_level) variable to the clients instead of handling it
> server-side. If a user now wants to change the setting, he'll have to
> reboot his phone after doing so. Changing the
> reg.x.callsPerLineKey setting in the phone also limits the number of
> outgoing calls the user can make. We'll also have to make code for all
> the other hardphones the customer is planning to use plus make guides
> on how to change the setting for all kinds of softphones. We want to
> avoid all this.
> With kind regards,
> Pan
>
> ----- Original Message -----
> *From:* Daniel-Constantin Mierla <mailto:miconda@gmail.com>
> *To:* Pan B. Christensen <mailto:pan@ibidium.no>
> *Cc:* sr-users(a)lists.sip-router.org
> <mailto:sr-users@lists.sip-router.org>
> *Sent:* Monday, December 13, 2010 12:26 PM
> *Subject:* Re: [SR-Users] Dialog module with 2 servers and 2
> separate databases.
>
>
>
> On 12/10/10 2:17 PM, Pan B. Christensen wrote:
>> Hello,
>> My customer has the following database design.
>> Voip server 1 talks to SQL server 1.
>> Voip server 2 talks to SQL server 2.
>> Voip 1 and Voip 2 are load-balanced.
>> Each SQL server has two databases. Database 1 contains
>> semi-static data like call forwarding properties for users and is
>> read-only. This is replicated from a third SQL server which the
>> web interface writes to. Database 2 is read/write, is not
>> replicated and contains data that is updated frequently like user
>> location and now dialog info.
>> Voip server 1 is not allowed to talk to SQL server 2 and vice versa.
>> I'm using forward() to send authenticated REGISTERs to the other
>> server so that it'll write this to RAM and its own SQL server.
>> Thus, both servers are aware of clients authenticated and
>> registered by the other server.
>> How can I make both servers be aware of active calls on the other
>> server?
> what is the purpose?
>
> Practically, it is not possible to track a call in two instances,
> because, unlike registration where is just a storage of mappings
> between contact and aor, call states of dialog module involve more
> processing logic, including timeouts and sending BYEs.
>
> Cheers,
> Daniel
>
> --
> Daniel-Constantin Mierla
> Kamailio (OpenSER) Advanced Training
> Jan 24-26, 2011, Irvine, CA, USA
> http://www.asipto.com
>
> ------------------------------------------------------------------------
>
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> _______________________________________________
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--
Daniel-Constantin Mierla
Kamailio (OpenSER) Advanced Training
Jan 24-26, 2011, Irvine, CA, USA
http://www.asipto.com
