[OpenSIPS-Users] Call pickup

[Bogdan-Andrei Iancu]

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.
>
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