How to handle incoming fragmented INVITEs on Asterisk?

Question

Asterisk

All the best!
Background:
there was a platform on which Asterisk 11.7 was assembled from sorts with all the consequences (that is, what was installed there a year ago will remain a mystery). External numbers from customers landed on this platform.
Among these numbers, there was one from Rostelecom (as it turned out in the end) with external registration and 4 lines for multi-channel. It was all set up and working great. After some time, the numbers needed to be transferred to another platform, and then miracles began.
Problem:
after setting up a trunk on another server, incoming calls to this number stopped coming! After a two-way check with the provider, it turned out that he (the provider) sends incoming INVITEs in 2 frames (ie > 1500b), which was visible from the log.
Because call forwarding by number on the provider’s side is set to SIP registration and to the SIP phone that the client has in the office, then the call, after several unsuccessful attempts to get a SIP 200 Ok response from us, was sent to the client’s phone.
Moreover, if you end the call before the handset is picked up, then we see two packets:
CANCEL and SIP / 2.0 481 Call leg / transaction does not exist
Which, in principle, is logical, because. these packets are not fragmented.
Accordingly, packets with authorization pass with a bang.
The situation was simulated on 3 servers with Asterisk 11.7 installed from the repository and FreePBX. Neither there nor there does the asterisk see the invite! Although on the platform on which it was originally configured - everything works! SIP settings are essentially identical. The only difference is in the assembly of the aster itself.
The problem is definitely not in the firewall, because. no rules have been configured. The servers were on different hostings, so I exclude the problem of dropping the 2nd frame at the network level.
Question:
Maybe someone faced a similar problem - what exactly is missing in the system for Asterisk to understand fragmented invites?
PS: unfortunately the old platform has gone to waste and it is no longer possible to restore or see what was collected there :(

Answer 1

And it is impossible to reduce the packet size, what would fit into 1? Remove the whole set of codecs, leave 1. What is this INVITE more than 1500 Oo?

Answer 2

options:
1. try to register via TCP
2. if the uplink for SIP is separate, increase the MTU of the interfaces and on all intermediate "too smart" switches