Why can the connection be broken?

Question

Asterisk

Given:
Provider Rostelecom SIP
PBX Yeastar S20
Cisco SPA504G Phones Akk is
registered with the provider, the phone is registered with the PBX.
When calling, both incoming and outgoing, the connection breaks after 186 seconds.
I rummaged through the PBX settings, there is no clear indication of such timeouts.
Tomorrow we are scheduled to communicate with the TP RT together we will watch the VIP log.
A question for connoisseurs, perhaps just in time you will have an idea in which direction to look.
Perhaps with some kind of re-polling, the connection is broken, but only the log will tell in detail.
No one is silent when speaking.

Answer 1

Do you have Wireshark disabled?

Answer 2

Go to S20 via SSH, run 2 instances of tcpdump there - on the external interface towards the operator and on the internal one, towards phones. Something like this: tcpdump -i eth0 -n -s0 -w /tmp/dump-eth0.pcap
It's also a good idea to run asterisk -rvvvvv to collect data from the console. Just in case.
Next, make a test call, wait for the "break" after 186 seconds. Interrupt tcpdump, upload both dumps to your desktop, open both with the Wireshark already mentioned above, see what happened (the Telephony menu - Sip flows will help a lot).
The question is "in time" - 186 seconds is too much to not correspond to some SIP timings for interrupting a conversation. Usually 15 or 30 seconds - lost "ACK". And 186 - rather, the duration of the call is erroneously set somewhere. Maybe NAT loses "translation", but then the voice (RTP) would be interrupted, and the call itself would continue.

Answer 3

Turn off noise cancellation on your phone