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 , [email protected] 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 [email protected]> 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, [email protected] 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 [email protected]> 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
>
> >
>
> > [email protected]
>
> >
>
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> >
>
> >
>
> >
>
> >
>
> >
>
>
>
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