[OpenSIPS-Users] How to tear down a call, when the UA receives a new INVITE? Is it possible?

[Rodrigo Pimenta Carvalho]

Dear OpenSIPS-users,


Let´s suppose someone (S1) is talking at a softphone.

Let´s suppose another person (S2) needs to talk to S1 at same time.

So, S2 calls S1.


How to configure OpenSIPS to tear down ("hang up") the current S1 call and then  send the S2 INVITE message to S1? Some example?


In my scenery, the user in S2 has a kind of high priority when calling any another one. So, S2 must be answered as soon as possible.


Any hint will be very helpful!


Best regards.

?


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
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