[OpenSIPS-Users] Call pickup

duane.larson at gmail.com duane.larson at gmail.com
Mon Jun 11 19:08:58 CEST 2012


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.



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