Call drop when SIP display info present

Hello.

My Asterisk drop some call from trunk. When I tcpdump packets I found that Asterisk incorrect parsing SIP packet. It said “No closing bracket found…”, but bracket realy exists.
Check screens:

I remove display info from other side and all works well.
How to fix this incorrect parsing?

Looks like it is too long and got truncated.

Note this is not the bug reporting system.

In my view, the peer is being unreasonable.