Asterisk, PJSIP how to understand why 503 error?

Question

Asterisk

Good afternoon, I know the asterisk from the book In the 5th chapter, I started 2 PJSIP points through the database. Everything is authorized without problems.
Simple dialplan:

[general]
[globals]
[sets]
exten => 100,1,Dial(PJSIP/0000f30A0A01)
exten => 101,1,Dial(PJSIP/0000f30B0B02)
exten => 102,1,Dial(PJSIP/SOFTPHONE_A)
exten => 103,1,Dial(PJSIP/SOFTPHONE_B)
exten => 200,1,Answer()
 same => n,Playback(hello-world)
 same => n,Hangup()

here is my pjsip.conf config

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0
[transport-tls]
type=transport
protocol=tls
bind=0.0.0.0
cert_file=/home/asterisk/certs/self-signed.crt   
priv_key_file=/home/asterisk/certs/self-signed.key

When I register on the server 2 points with the "transport-tls" transport: softphone (microsip) and zoiper (on a mobile phone) - calls successfully go only to a mobile phone. no calls from mobile zoiper to softphone. Here is what pjsip show history



gives out. A similar situation is if I register instead of a microsip softphone - a regular Yealink SIP-T19P voip phone.

If you register both subscribers from mobile phones (zoiper), then calls go in two directions without problems.

Here is what is in the asterisk console:



And with all this, if I take one of the subscribers - the subscriber with the transport [transport-udp], then everything works.

asterisk 16.15.0 in virtualBOX, network bridge, centos7 OS, codecs ulaw, SELinux on Permissive

where to dig, how and what logs to watch?

Thanks