Why is the incoming call dropped?

Question

Asterisk

Good afternoon! I'm trying to set up a SIP trunk, registration is established, outgoing go, incoming are reset.
I just can’t understand what’s the matter, I set up a sip account on an Ip phone, with the same settings, everything works.
When I turn on the trunk, the incoming ones are dropped.
What's in the logs:
SIP/2.0 401 Unauthorized Via: SIP /
2.0/UDP 31.44.92.216:5060;branch=z9hG4bK-524287-1---14d2097bc70e97f5c740fb9e50586c8a;received=31.44.92.206;rportia=31.44.92.206;rportia=31.44.92.206;rportia
UDP 31.44.92.217:5071;rport=5071;branch=z9hG4bK-javhnlpp4jbefdpn
From: <<

Answer 1

Dump the call with tcpdump -i eth0 'port 5060' -s0 -w /tmp/dump.pcap
and see. You can post it here for group discussion.
A possible option (one of) - you register on the sip.example.com server by FQDN (by name), and this name is used in SIP packets, but an incoming call comes to you with a different address in the SIP packet, and asterisk does not identify this incoming call with a configured peer.
In this case, it will help to register by IP address, and not by FQDN.