[OpenSIPS-Users] Call pickup

[Bogdan-Andrei Iancu]

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
> >
> >
> >
> >
> >