[OpenSIPS-Users] Call pickup

Tue Jun 12 02:57:47 CEST 2012

Well using "get_dialog_info" works but on the Snom phone it won't ring  
since initially you are already on a call. Then the Call is canceled and  
you see that a second line is receiving a call. So with you select that  
second line you are connected to CallerA. Its a dirty hack but works. The  
only issue I see is when CallerA hangs up CallerC still thinks he is on the  
call and the call doesn't terminate. Will have to see whats going wrong  
there.

I will still try the AVP way and see if that is a cleaner hack.

On , duane.larson at gmail.com wrote:
> Yeah it makes a little sense. I guess I would need to do a MySQL DB query  
> and add the URI into an AVP.

> I found this post on Nabble and thought I might be able to use it
> http://opensips-open-sip-server.1449251.n2.nabble.com/NEW-exchanging-info-between-dialogs-td4975220.html#a5035451

> I will test to see if this works. Here is what I am thinking

> C sends a Call Pickup INVITE to PREFIX+A.

> OpenSIPS does the following when it gets this INVITE
> if(search("^Replaces:.*;")){
> # Grab the CallID in the Replaces Header so we can cancel the call to  
> User B
> $var(Replacesb2b) = $(hdr(replaces){s.select,0,;});
> exec_msg("/usr/local/sbin/opensipsctl fifo t_uac_cancel $var(Replacesb2b)  
> 2");

> # - Set the dialog variables so the B2B dialog can see who we need to  
> fail the call over to
> store_dlg_value("CallPickupGrabber","$tU"); # - Set the dialog variables  
> so the B2B dialog can see what CallID to Cancel
> store_dlg_value("CallPickupCallID","$ci");
> # - Set the value of the new URI the call needs to go to when it fails  
> over
> store_dlg_value("CallPickupNewCallee","$fu"); };

> So a Cancel gets sent to B

> Now the first dialog, the B2B dialog, goes to failure route and we need  
> to do the following within the failure_route

> if(get_dialog_info("CallPickupCallID","$var(x)","CallPickupGrabber","$fU")  
> )  
> { $dlg_val("CancelCall") = $var(x)
> }

> if(get_dialog_info("CallPickupNewCallee","$var(y)","CallPickupGrabber","$fU")  
> )  
> { $dlg_val("NewBranch") = $var(y)
> }

> # Cancel the call from Caller C who wanted to do a call pickup
> exec_msg("/usr/local/sbin/opensipsctl fifo t_uac_cancel  
> $dlg_val("CancelCall") 1");

> # Set the new Branch call
> $ru = "sip:" + $dlg_val("NewBranch");

> t_relay();
> exit;

> So I would think that Caller C will press a softkey when he wants to do a  
> CallPickup and by pressing the key the call should be canceled and then  
> Caller C's phone will ring and he can then talk to Caller A

> Not sure if my logic is correct or if the get_dialog_info will solve my  
> problems. Any thoughts on if you think this might work or not (I'll have  
> to test later)?

> On , Bogdan-Andrei Iancu bogdan at opensips.org> wrote:
> > Well, this is indeed a missing piece - some kind of way to pass  
> information between transactions - either directly append a new branch  
> for another transaction (based on AVP matching ?), either a more generic  
> way to add an AVP to another transaction.
> >
> >
> >
> > Does it make sense ?
> >
> >
> >
> > Regards,
> >
> >
> >
> > Bogdan-Andrei Iancu
> >
> > OpenSIPS Founder and Developer
> >
> > http://www.opensips-solutions.com
> >
> >
> >
> >
> >
> > On 06/11/2012 08:08 PM, duane.larson at gmail.com wrote:
> >
> >
> > I was thinking about this last night and with the Snom Phones I am able  
> to do the following.
> >
> >
> >
> > I know how to cancel the first call to the PhoneB and I am thinking  
> that I can cancel the call that PhoneC does when it wants to do a Call  
> Pickup. The thing I am not sure about is how on the first call to add a  
> branch to PhoneCs URI. How can I pass PhoneC's URI info to the first call  
> so that I can add the new branch?
> >
> >
> >
> > On , Bogdan-Andrei Iancu bogdan at opensips.org> wrote:
> >
> > > Hi Duane,
> >
> > >
> >
> > >
> >
> > >
> >
> > > How I see this "call pickup" functionality:
> >
> > >
> >
> > >
> >
> > >
> >
> > > 1) A calls to B, call is in ringing state
> >
> > >
> >
> > >
> >
> > >
> >
> > > 2) C wants to pickup ringing call to B (this means C want to get to  
> his phone the the call ringing from B).
> >
> > >
> >
> > >
> >
> > >
> >
> > > 3) C dials PREFIX+B, indicating he wants to grab the call for B)
> >
> > >
> >
> > >
> >
> > >
> >
> > > 4) the INVITE for (for the call from C) should add a new branch to C  
> ( for the call to B) and to cancel the branch to B
> >
> > >
> >
> > >
> >
> > >
> >
> > > 5) as a result, the call from C will be terminated and the call from  
> A will be serially forked to C.
> >
> > >
> >
> > >
> >
> > >
> >
> > > This is how I see this scenario.
> >
> > >
> >
> > >
> >
> > >
> >
> > > Now there are same small missing pieces to make this happen - the  
> most important is first to decide if the manipulation over the first call  
> (adding a new branch and terminating the ongoing branch) should be done  
> from script or via MI.
> >
> > >
> >
> > >
> >
> > >
> >
> > > Regards,
> >
> > >
> >
> > >
> >
> > >
> >
> > > Bogdan-Andrei Iancu
> >
> > >
> >
> > > OpenSIPS Founder and Developer
> >
> > >
> >
> > > http://www.opensips-solutions.com
> >
> > >
> >
> > >
> >
> > >
> >
> > >
> >
> > >
> >
> > > On 06/10/2012 03:44 AM, osiris123d wrote:
> >
> > >
> >
> > >
> >
> > > Bogdan,
> >
> > >
> >
> > >
> >
> > >
> >
> > > I'm trying to figure out how to get Call Pickup working since the PSTN
> >
> > >
> >
> > > provider can't handle the Replaces: header. Here is my post here
> >
> > >
> >
> > >
> >
> > >
> >
> > >  
> http://opensips-open-sip-server.1449251.n2.nabble.com/B2B-with-Call-Pickup-td7580224.html
> >
> > >
> >
> > >
> >
> > >
> >
> > > I see in this post you talk about using MI commands and the TM and  
> Dialog
> >
> > >
> >
> > > modules and the failure route to make this work. I think with the TM  
> module
> >
> > >
> >
> > > I can send a CANCEL to the original Callee but how would you make the  
> call
> >
> > >
> >
> > > then fail over to the Failure Route so I can send it to the next  
> callee?
> >
> > >
> >
> > >
> >
> > >
> >
> > > --
> >
> > >
> >
> > > View this message in context:  
> http://opensips-open-sip-server.1449251.n2.nabble.com/Call-pickup-tp7127393p7580251.html
> >
> > >
> >
> > > Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
> >
> > >
> >
> > >
> >
> > >
> >
> > > _______________________________________________
> >
> > >
> >
> > > Users mailing list
> >
> > >
> >
> > > Users at lists.opensips.org
> >
> > >
> >
> > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> > >
> >
> > >
> >
> > >
> >
> > >
> >
> > >
> >
> >
> >
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