[OpenSIPS-Users] Call Pickup Feature

[Duane Larson]

I think there might be a little confusion here and a language barrier.  If
I am understanding Jorge correctly I think he is stating that OpenSIPS is
load balancing between many Asterisk servers.  OpenSIPS can do the Call
Pickup feature as can Asterisk.  I think the issue is that if OpenSIPS is
doing Load balancing and UAC1 is ringing and UAC2 wants to "Call Pickup"
that call then OpenSIPS needs to route UAC2's request to the same Asterisk
server that UAC1 was called on.

Hope I understood Jorge correctly.  If I did then you would need to work
with "get_dialog_info" function from the Dialog Module (
http://www.opensips.org/html/docs/modules/1.10.x/dialog.html#id295324)

So you would need to do something like this

        if ( get_dialog_info("host","$var(x)","caller","$fU") ||
get_dialog_info("host","$var(x)","caller","$rU") ) {

                #route to host $var(x)

                $du = "sip:" + $rU + "@" + $var(x) + ":5060";

                if ( !match_dialog() )

                {

                        create_dialog();

                }

                $dlg_val(host) = $du;

        } else if ( get_dialog_info("host","$var(x)","caller","$rU") ||
get_dialog_info("host","$var(x)","caller","$fU") ) {

                #route to host $var(x)

                $du = "sip:" + $rU + "@" + $var(x) + ":5060";

                if ( !match_dialog() )

                {

                        create_dialog();

                }

                $dlg_val(host) = $du;

        } else {


                if ( !match_dialog() )

                {

                        create_dialog();

                }

                $dlg_val(caller) = $fU;

                $dlg_val(callee) = $rU;

        }



You will need to somehow make this work for your setup but hopefully this
shows you what you are looking for.


On Mon, Dec 16, 2013 at 7:31 PM, Jeff Pyle <jpyle at fidelityvoice.com> wrote:

> Jorge,
>
> This is a function of Asterisk, not Opensips.  This page may help you:
>   http://www.voztovoice.org/?q=node/350
>
>
> - Jeff
>
>
>
> On Mon, Dec 16, 2013 at 7:00 PM, Jorge Ortea <darham at hotmail.com> wrote:
>
>> Hi all,
>>
>> Suppose a platform with OpenSIPS and several Asterisk behind. A new call
>> in a Asterisk that send to Opensips to route to uac1. The uac1 is ringing,
>> it is sending 180 Ringing, then from other uac wants CallPickup this call,
>> this feature is dialed but when the Invite reach to OpenSIPS,,, How I can
>> know that Asterisk is the call?
>>
>>
>> Very Thanks.
>> Regards.
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


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Duane
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