Asterisks - SIP reinvites discards frames from jitter buffer

We are testing our system (Asterisks-Client - SIP-Server - Asterisks Client) with a ISDN bit error rate tester.
We get for each reinvite from the SIP-Server to the Asterisks Client bit errors. The
client responds correctly but discards frames from the jitter buffer.
Code reading shows that asterisks stops the bridge and resets the jitter buffer.
How can we solve this issue? Is this behaviour correct?
We are using Asterisk 11. Is there a version which does not discards the frames from the jitter buffer?

Dear Christian,

As per your concern you are facing the issue of jitter.Request you please confirm that Is you install Asterisk on virtual machine or physical machine.

Asterisks is installed on a physical machine.