Hello,
Try something like this :
if ( get_dialog_info("host","$var(x)","caller","$fU") ) {
xlog("caller $fU has another ongoing, on host $var(x)\n")
#route to host $var(x)
} else if ( get_dialog_info("host","$var(x)","callee","$rU") ) {
xlog("callee $rU has another ongoing, on host $var(x)\n")
#route to host $var(x)
} else {
create_dialog();
$dlg_val(caller) = $fU;
$dlg_val(callee) = $rU;
load_balance("1","pstn");
$dlg_val(host) = $du;
}
Regards,
Vlad Paiu
OpenSIPS Developer
On 12/04/2011 04:25 PM, Nick Khamis wrote:
Schneur,
I'm running into this problem now as well. Do you want to work on
this together?
Ninus.
On Thu, Dec 1, 2011 at 1:04 PM, Schneur Rosenberg
<[email protected]> wrote:
Bogdan there is too little info about this online, can you please help
me a bit more with this, how do I write the if statement, and how do I
set a variable for the first call, and how do I retrieve which server
was used for the first call.
On Wed, Nov 23, 2011 at 9:46 PM, Schneur Rosenberg
<[email protected]> wrote:
thank you Bogdan
On Wed, Nov 23, 2011 at 7:32 PM, Bogdan-Andrei Iancu
<[email protected]> wrote:
Hi Schneur,
What you have to do is to change the way you distribute the call among the
asterisk boxes in such a way that all calls in which a user is involved to
be on the same box (so that the transfers will work).
How to do that? with a mixed routing logic. When you receive a new call, do:
- check if caller or callee are already involved into an existing call on
a certain box. if so, route to that box
- default is to do LB as you do now.
For the check part, you need to use the dialog module (to be dialog
stateful), set in some dialog variables the caller / callee / box (to be
remembered later) and query via get_dialog_info() function -
http://www.opensips.org/html/docs/modules/1.7.x/dialog.html#id294051
Regards,
Bogdan
On 11/23/2011 06:48 PM, Schneur Rosenberg wrote:
I'm using Opensips as a Load balancer and as a registrar, so basically
all phones are registered to the Opensips, all Incoming calls hit the
opensips server which forwards the call to asterisk with load
balancing, asterisk decides what to do with the call ie IVR voicemail
etc and if the call needs to be sent to a phone asterisk will send it
back to opensips and opensips will send it to the phone.
Outgoing calls are sent to asterisk via load balancing and asterisk
decides how to terminate the call.
This setup helps me load balance all calls and also removes the
registrar load from asterisk which does not handle registrations fine
when there are approx 300 peers on my asterisk system.
My problem is that sometimes when I do a transfer I get back from
asterisk "SIP/2.0 481 Call leg/transaction does not exist.".
The test call I've done was done by calling from phone 1 a phone
number which hits our system, so what happened is phone invited
opensips to the DID, opensips sent the call to Asterisk server 1, then
the DID called in and opensips sent it to Asterisk server 2, Asterisk
server 2 saw that this did should ring on a phone so it sent it back
to opensips which properly terminated the call to phone 2, then phone
1 wanted to transfer call to a outside phone, so it sent a invite to
opensips with the phone number to call, opensips sent call to Asterisk
server 2, then when user on phone 1 hit transfer, phone sent a refer
to Asterisk 1, and asterisk 1 retuned a NOTIFY with
Subscription-state: terminated;reason=noresource. and SIP/2.0 481 Call
leg/transaction does not exist.
Can anyone please help me solve this problem.
thank you
S. Rosenberg
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--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
OpenSIPS solutions and "know-how"
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http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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