Why is the incoming call not going through (Elastix)?

Question

Asterisk

The provider gives a trunk on IP without registration. Outgoing configured, everything works. Incoming problem.
10.X.242.42 - Asterisk IP address
62.X.15.8 - Trunk gateway IP address

[2016-02-10 21:55:50] VERBOSE[3255] chan_sip.c: Reliably Transmitting (no NAT) to 62.X.15.8:5060:
OPTIONS sip:62.X.15.8 SIP/2.0
Via: SIP/2.0/UDP 10.X.242.42:5060;branch=z9hG4bK2a4d99ff
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as1499ba48
To: <sip:62.X.15.8>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.17.1)
Date: Wed, 10 Feb 2016 20:55:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[2016-02-10 21:55:50] VERBOSE[3255] chan_sip.c: 
<--- SIP read from UDP:62.X.15.8:5060 --->
SIP/2.0 480 No Routes Found
Via: SIP/2.0/UDP 10.X.242.42:5060;branch=z9hG4bK2a4d99ff
From: "Unknown" <sip:[email protected]>;tag=as1499ba48
To: <sip:62.X.15.8>;tag=aprqngfrt-ng5ugi20000c6
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS

<------------->
[2016-02-10 21:55:50] VERBOSE[3255] chan_sip.c: --- (6 headers 0 lines) ---
[2016-02-10 21:55:50] VERBOSE[3255] chan_sip.c: Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
[2016-02-10 21:55:51] VERBOSE[3255] chan_sip.c: Reliably Transmitting (no NAT) to 62.X.15.8:5060:
OPTIONS sip:62.X.15.8 SIP/2.0
Via: SIP/2.0/UDP 10.X.242.42:5060;branch=z9hG4bK5a33b1ea
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as06822043
To: <sip:62.X.15.8>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.17.1)
Date: Wed, 10 Feb 2016 20:55:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[2016-02-10 21:55:51] VERBOSE[3255] chan_sip.c: 
<--- SIP read from UDP:62.X.15.8:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.X.242.42:5060;branch=z9hG4bK5a33b1ea
From: "Unknown" <sip:[email protected]>;tag=as06822043
To: <sip:62.X.15.8>;tag=snl_0011502104
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Contact: <sip:[email protected]:5060;transport=udp;maddr=62.X.3.76>
Supported: sip-stun
Content-Length: 0

<------------->
[2016-02-10 21:55:51] VERBOSE[3255] chan_sip.c: --- (9 headers 0 lines) ---
[2016-02-10 21:55:51] VERBOSE[3255] chan_sip.c: Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
[2016-02-10 21:55:50] VERBOSE[3255] chan_sip.c: Reliably Transmitting (no NAT) to 62.X.15.8:5060:
OPTIONS sip:62.X.15.8 SIP/2.0
Via: SIP/2.0/UDP 10.X.242.42:5060;branch=z9hG4bK2a4d99ff
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as1499ba48
To: <sip:62.X.15.8>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.17.1)
Date: Wed, 10 Feb 2016 20:55:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[2016-02-10 21:55:50] VERBOSE[3255] chan_sip.c: 
<--- SIP read from UDP:62.X.15.8:5060 --->
SIP/2.0 480 No Routes Found
Via: SIP/2.0/UDP 10.X.242.42:5060;branch=z9hG4bK2a4d99ff
From: "Unknown" <sip:[email protected]>;tag=as1499ba48
To: <sip:62.X.15.8>;tag=aprqngfrt-ng5ugi20000c6
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS

<------------->
[2016-02-10 21:55:50] VERBOSE[3255] chan_sip.c: --- (6 headers 0 lines) ---
[2016-02-10 21:55:50] VERBOSE[3255] chan_sip.c: Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
[2016-02-10 21:55:51] VERBOSE[3255] chan_sip.c: Reliably Transmitting (no NAT) to 62.X.15.8:5060:
OPTIONS sip:62.X.15.8 SIP/2.0
Via: SIP/2.0/UDP 10.X.242.42:5060;branch=z9hG4bK5a33b1ea
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as06822043
To: <sip:62.X.15.8>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.17.1)
Date: Wed, 10 Feb 2016 20:55:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[2016-02-10 21:55:51] VERBOSE[3255] chan_sip.c: 
<--- SIP read from UDP:62.X.15.8:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.X.242.42:5060;branch=z9hG4bK5a33b1ea
From: "Unknown" <sip:[email protected]>;tag=as06822043
To: <sip:62.X.15.8>;tag=snl_0011502104
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Contact: <sip:[email protected]:5060;transport=udp;maddr=62.X.3.76>
Supported: sip-stun
Content-Length: 0

<------------->
[2016-02-10 21:55:51] VERBOSE[3255] chan_sip.c: --- (9 headers 0 lines) ---
[2016-02-10 21:55:51] VERBOSE[3255] chan_sip.c: Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

A few questions along the way.
1. Why is there no INVITE from the server? Shouldn't the incoming call be initialized from the gateway, in our case from 62.X.15.8 ?
2. Why is first defined as

From: "Unknown" <sip:[email protected]>;tag=as1499ba48

and then how

From: "Unknown" <sip:[email protected]>;tag=as06822043.

Why is the phone number not immediately determined?
3. Do I understand correctly that

[2016-02-10 21:55:50] VERBOSE[3255] chan_sip.c: Reliably Transmitting (no NAT) to 62.X.15.8:5060:
OPTIONS sip:62.X.15.8 SIP/2.0
Via: SIP/2.0/UDP 10.X.242.42:5060;branch=z9hG4bK2a4d99ff
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as1499ba48
To: <sip:62.X.15.8>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.17.1)
Date: Wed, 10 Feb 2016 20:55:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

means that "Unknown" <sip:[email protected]>sends information for <sip:62.X.15.8>?

Answer 1

Behind nat you have an asterisk. You will have to hang it on an external IP and configure IPtables

Answer 2

I called the provider. They changed the configuration after which everything worked.

Answer 3

On the router, you need to do the following:
1. Set up port forwarding from your provider's network to the PBX host 2.
Try to disable Sip helper or SIP ALG, sometimes they interfere