Linphonec, unable register extension to freePBX 15

Hi Guys.
I need some help.

I have project using two orangepi.
On one of them is installed freePBX 15

I need:
Register some cli softphone installed on server, and make call to conference room
And register cli softphone on the second orangepi to the server freePBX on the first one, and make call to the same conference room.
Everything has to happen automatically after system boot.
When both devices connect, after inserting headphones with a microphone into the orangespi adio ports, I should have an active channel between the both orangepi.

FreePBX works properly, i have installed linphonec but i cant to register neither to sip nor pjsip.
when trying to connect, asterisk shows absolutely nothing in the logs, but linphonec shows:

linphonec> register sip:[email protected]:5160 1234
2023-05-31 16:40:00:698 liblinphone-message-linphone_core_find_auth_info(): returning auth info username=10, realm=
2023-05-31 16:40:00:699 liblinphone-message-linphone_proxy_config_is_server_config_changed : 1
2023-05-31 16:40:00:699 liblinphone-message-Publish params have not changed on proxy config [0x55c027bdb0]
linphonec> register sip:[email protected]:5060 1234
2023-05-31 16:40:29:874 liblinphone-message-linphone_core_find_auth_info(): returning auth info username=100, realm=
2023-05-31 16:40:29:874 liblinphone-message-linphone_proxy_config_is_server_config_changed : 0
2023-05-31 16:40:29:874 liblinphone-message-Publish params have changed on proxy config [0x55c027bdb0]

Exten 10 is SIP
Exten 100 is PJSIP

On SIP i changed:


There is no such option (and nat=no is rarely needed - mainly just Cisco).

You need to provide Asterisk logs, probably with both “sip set debug on” and “pjsip set logger on” enabled.

I suggested this: Learn how to Use Asterisk with a Linphone Softphone
As i said in asterisk cli no any reaction on those tries

At the same time, 99 of jitsi is registered on windows, which works fine:

<--- Transmitting SIP request (454 bytes) to UDP: --->
OPTIONS sip:[email protected]:54308;rinstance=1ab57e3980fd727e SIP/2.0
Via: SIP/2.0/UDP;rport;branch=z9hG4bKPj028e2fb6-822d-4df4-bc80-8f4dd855c654
From: <sip:[email protected]>;tag=a9f63e25-93b9-4c66-b718-94a6b08b0b3a
To: <sip:[email protected];rinstance=1ab57e3980fd727e>
Contact: <sip:[email protected]:5060>
Call-ID: 9ffe6207-2bb0-4e95-8734-f297131eabbd
CSeq: 29817 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-15.0.23(18.10.0)
Content-Length:  0

<--- Received SIP response (579 bytes) from UDP: --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP;rport=5060;branch=z9hG4bKPj028e2fb6-822d-4df4-bc80-8f4dd855c654
Contact: <sip:>
To: <sip:[email protected];rinstance=1ab57e3980fd727e>;tag=de5c4078
From: <sip:[email protected]>;tag=a9f63e25-93b9-4c66-b718-94a6b08b0b3a
Call-ID: 9ffe6207-2bb0-4e95-8734-f297131eabbd
CSeq: 29817 OPTIONS
Accept: application/sdp
Accept-Language: en
Supported: replaces
Allow-Events: presence, message-summary, tunnel-info
Content-Length: 0

If you are not getting anything logged, the requests aren’t getting to Asterisk.

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