SIP Problem (Asterisk)

I am having issue with SIP call forking in Asterisk and the issue is

As soon as I received the call from PSTN/trunk. I am sending it to all agents using Dial command

now the problem is
The single call is attended by two agents at the same time and causing confusion.

Is there any specific SIP settings that allow to drop 200 ok message. For the same call session ID or CSeq id.

Asterisk is not a proxy. Each call has a different callid, and, even if the cseq values happen to be the same, there is no significance in that.

Even in a proxy situation, one should not just ignore responses in unsuccessful branches.

Could you please provide details, with appropriate logging, Asterisk should close down any colliding calls. However, if they collide sufficiently precisely that both respond 200 OK, the SIP protocol requires that they be ACKed before they are closed with a BYE, so there will be a short SIP call for the failing contender.

In Asterisk, there is also a layering issue, in that the DIal application is running asynchronously with respect to the basic SIP response mechanism, so the ACK has probably been sent, even before Dial sees the collision.