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
_______________________________________________
Users mailing list
[email protected]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
