Sip calls are breaking in to each other

Hello friends

i have a softphone software that dial 3 calls at the same time and do predictive dialer.
the softphone is connected to a virtual asterisk online via the extension.
the problem is sometimes when i make a call and another user makes a call
after my call is answered the call from the other user also join into my call.
its like the sip call from extension 1 is breaking into extension 2 automaticly.

any idea what can cause these?

need more info from me?



Bugs in the soft phone software.

what can cause this bug?
is it the way the softphone connects to asterisk or the way the calls are being Assigned?

what shell i tell to my admin?