13 Dec
2010
13 Dec
'10
4:12 p.m.
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:
--
Daniel-Constantin Mierla
Kamailio (OpenSER) Advanced Training
Jan 24-26, 2011, Irvine, CA, USA
http://www.asipto.com
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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