I tried to do this simple task again.
And did as you advised.
But the result is
──────────┬───────── ──────────┬─────────│From: "oooooooo oooooooooooooo" <
sip:1...@172.17.3.31>;tag=fa1ae169-542c-40cb-a3a9-2062a3a9dc93
16:43:11.477786 │ INVITE (SDP) │ │To: <
sip:2...@172.17.1.28>
+0.001253 │ ──────────────────────────> │ │Contact: <
sip:aste...@172.17.3.31:5060>
16:43:11.479039 │ 404 Not Found │ │Call-ID: 17e44dad-fdd4-4c4c-bd87-f93625e6879b
+0.000479 │ <────────────────────────── │ │CSeq: 2568 INVITE
16:43:11.479518 │ ACK │ │Route: <sip:172.17.1.28;lr>
│ ──────────────────────────> │ │Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
│ │ │Supported: 100rel, timer, replaces, norefersub, histinfo
│ │ │Session-Expires: 1800
│ │ │Min-SE: 90
│ │ │Max-Forwards: 70
│ │ │User-Agent: FPBX-16.0.33(18.16.0)
│ │ │Content-Type: application/sdp
│ │ │Content-Length: 337
│ │ │
│ │ │v=0
│ │ │o=- 1922348612 1922348612 IN IP4 172.17.3.31
│ │ │s=Asterisk
│ │ │c=IN IP4 172.17.3.31
│ │ │t=0 0
│ │ │m=audio 14860 RTP/AVP 0 8 3 111 9 101
│ │ │a=rtpmap:0 PCMU/8000
│ │ │a=rtpmap:8 PCMA/8000
│ │ │a=rtpmap:3 GSM/8000
│ │ │a=rtpmap:111 G726-32/8000
│ │ │a=rtpmap:9 G722/8000
│ │ │a=rtpmap:101 telephone-event/8000
│ │ │a=fmtp:101 0-16
│ │ │a=ptime:20
│ │ │a=maxptime:150
│ │ │a=sendrecv
│ │ │
│ │ │
I never figured out how to make a route for two PBX stations