It's been a while, stumbled on this case just now. It seems like either nat_uac_test() or more likely fix_nated_sdp() doesn't catch the 192.0.0.0/29 subnet as private.
**Script logic:** route[NATMANAGE] { ... if (nat_uac_test("8")) fix_nated_sdp("15"); ... }
**Sipdump:** U 2020/04/07 10:36:02.572802 135.19.155.163:17669 -> 65.39.1.1:5060 INVITE sip:8007777777@client.mydomain.net:5060 SIP/2.0 Via: SIP/2.0/UDP 192.0.0.254:11198;branch=z9hG4bK1066155396 From: "ME" sip:2213@client.mydomain.net:5060;tag=3901872054 To: sip:8007777777@client.mydomain.net:5060 Call-ID: 6_327133011@192.168.0.78 CSeq: 1 INVITE Contact: sip:2213@192.0.0.254:11198 Content-Type: application/sdp Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE Max-Forwards: 70 User-Agent: Yealink SIP-T46S 66.84.0.10 Allow-Events: talk,hold,conference,refer,check-sync Supported: replaces Content-Length: 305
v=0 o=- 22668 22668 IN IP4 192.168.0.78 s=SDP data c=IN IP4 192.0.0.254 t=0 0 m=audio 22936 RTP/AVP 9 0 8 18 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=ptime:20 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15
U 2020/04/07 10:36:02.573384 65.39.1.1:5060 -> 135.19.155.163:17669 SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.0.0.254:11198;branch=z9hG4bK1066155396;rport=17669;received=135.19.155.163 From: "ME" sip:2213@client.mydomain.net:5060;tag=3901872054 To: sip:8007777777@client.mydomain.net:5060 Call-ID: 6_327133011@192.168.0.78 CSeq: 1 INVITE Server: NXO Content-Length: 0
U 2020/04/07 10:36:02.573608 65.39.1.1:5060 -> 66.199.2.2:5060 INVITE sip:8007777777@client.mydomain.net:5060 SIP/2.0 Record-Route: sip:65.39.1.1;lr;did=faf.517 Via: SIP/2.0/UDP 65.39.1.1;branch=z9hG4bK6ef8.b9248afe5375fc379eb33718de1f7481.0 Via: SIP/2.0/UDP 192.0.0.254:11198;rport=17669;received=135.19.155.163;branch=z9hG4bK1066155396 From: "ME" sip:2213@client.mydomain.net:5060;tag=3901872054 To: sip:8007777777@client.mydomain.net:5060 Call-ID: 6_327133011@192.168.0.78 CSeq: 1 INVITE Contact: sip:2213@192.0.0.254:11198;alias=135.19.155.163~17669~1 Content-Type: application/sdp Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE Max-Forwards: 69 Allow-Events: talk,hold,conference,refer,check-sync Supported: replaces Content-Length: 281 .. v=0. o=- 22668 22668 IN IP4 192.168.0.78 s=SDP data. c=IN IP4 192.0.0.254 t=0 0 m=audio 22936 RTP/AVP 9 0 18 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=ptime:20 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15