Hi Duane,
A work around will be:
1) you receive the call from C (doing the pickup)
2) dig for original call from A to B
3) close the call to C by sending a negative reply
4) continue processing in script (for C call) by pushing the new branch
for call A to B
5) put 1 sec delay
6) cancel the original branch for A to B
This chaining will guarantee that you will get (on C) first hanged the
outgoing call and only after that you will receive the can call from A .
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 06/12/2012 03:57 AM, [email protected] 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 , [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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