Why does Asterisk not see trunks?

Question

Asterisk

Hello everyone,
The point is:
*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
0 SIP registrations.
sip_registrations
register=userX:[email protected]
Here is the debug

Reliably Transmitting (no NAT) to 81.88.86.11:5060:
OPTIONS sip:81.88.86.11 SIP/2.0
Via: SIP/2.0/UDP 185.60.134.230:5060;branch=z9hG4bK511b0208
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as34b5b626
To: <sip:81.88.86.11>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-12.0.76.2(13.5.0)
Date: Mon, 26 Oct 2015 18:19:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[2015-10-26 21:19:55] DEBUG[2764]: chan_sip.c:3709 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for 81.88.86.11:5060
[2015-10-26 21:19:55] DEBUG[2719]: threadpool.c:508 grow: Increasing threadpool stasis-core's size by 1
[2015-10-26 21:19:55] DEBUG[3116]: taskprocessor.c:484 tps_taskprocessor_destroy: destroying taskprocessor '7109f8b0-7c47-4d87-926f-a78bf2aa18ed'

<--- SIP read from UDP:81.88.86.11:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 185.60.134.230:5060;received=185.60.134.230;rport=5060;branch=z9hG4bK511b0208
From: "Unknown" <sip:[email protected]>;tag=as34b5b626
To: <sip:81.88.86.11>;tag=e7d2c81e2d36b140069cf4a65dc24be5.d0b6
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Server: Mango SIP
Content-Length: 0

Answer 1

Or maybe try to apply sip.conf for this?
In sip.conf under [general] add a register definition:
Format:
register => user[:secret[:authuser]]@host[:port][/extension]

Answer 2

Isn't it because the register string in Asterisk must have the format register=>userX:[email protected]
(there is not = , but => )?

Answer 3

It’s quite an option that the cant is in the still very fresh version of FreePBX.
Try to just specify the callbackextension parameter in the trunk settings
....
type=friend
callbackextension=FOR THE CITY TELEPHONE
...
In the inbound routes, set the DID equal to the callbackextension value
What you wrote in the SIP settings is a crutch, when you change / delete the trunk, you can forget this crutch put away