[OpenSIPS-Users] Call pickup
duane.larson at gmail.com
duane.larson at gmail.com
Tue Jun 12 04:57:04 CEST 2012
Just tested the AVP method and that is a lot cleaner since CallerC's INVITE
with the Replaces header gets canceled way before the failure_route sets
the new branch to CallerC. This method also doesn't give me the issue that
I was seeing in the last method where CallerA hung up and CallerC still
thought CallerA was on the phone.
Here is what I did
RouteX
if(search("^Replaces:.*;")){
$var(Replacesb2b) = $(hdr(replaces){s.select,0,;});
exec_msg("/usr/local/sbin/opensipsctl fifo t_uac_cancel $var(Replacesb2b) 2
> /tmp/test.txt");
t_reply("404", "All Good"); <----------- Can you think of anything better
to do here?
exit;
};
Failure_Route
if(avp_db_load("$ru/username","$avp(callpickupDest)"))
{
if ( $avp(callpickupDest) != null)
{
$rU = $avp(callpickupDest);
alias_db_lookup("dbaliases" , "d");
lookup("location");
t_relay();
exit;
}
}
I'm trying to figure out if there is a better way of using the IF statement
in RouteX instead of looking to see if there is a Replaces header. With
Snom phones I can't put a prefix in front of the number or else it wouldn't
be able to SUBSCRIBE for dialog updates and see if someone is calling the
person. I'm sure there is some way. Might just need to add a prefix and
when the SUBSCRIBE comes in strip it off.
Thanks for the help Bogdan!
On , duane.larson at gmail.com wrote:
> 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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