How to config SIP test local in Ubunto

[vtmjapandev (VTMDev)]

Hi guys,

As a new member, I’ve installed Asterisk and FreePBX and attempted to make calls using the Janus SIP plugin. However, I’m unable to register. The Asterisk logs show an error. Could you please help me troubleshoot this issue?

Screen Short FreePBX configuration(I can’t embeded multi media file Because a new member)

[2024-01-26 16:35:14] WARNING[2802216] res_pjsip_outbound_registration.c: No response received from 'sip:192.168.0.158:5060' on registration attempt to 'sip:[email protected]:5060', retrying in '60'
[2024-01-26 16:35:27] VERBOSE[2802216] res_pjsip_logger.c: <--- Transmitting SIP request (444 bytes) to UDP:192.168.0.158:5060 --->
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.9:5060;rport;branch=z9hG4bKPj69db6ce7-c623-49ac-b8bf-84a18741ceb9
From: <sip:[email protected]>;tag=ff7e15e5-d474-4fb3-a143-56d4bad629ee
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: 16b52af3-35a4-4e27-bfe8-1dfe3df9bb19
CSeq: 21607 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-16.0.40.7(20.5.2)
Content-Length:  0


[2024-01-26 16:35:28] VERBOSE[2802215] res_pjsip_logger.c: <--- Transmitting SIP request (444 bytes) to UDP:192.168.0.158:5060 --->
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.9:5060;rport;branch=z9hG4bKPj69db6ce7-c623-49ac-b8bf-84a18741ceb9
From: <sip:[email protected]>;tag=ff7e15e5-d474-4fb3-a143-56d4bad629ee
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: 16b52af3-35a4-4e27-bfe8-1dfe3df9bb19
CSeq: 21607 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-16.0.40.7(20.5.2)
Content-Length:  0


[2024-01-26 16:35:29] VERBOSE[2802215] res_pjsip_logger.c: <--- Transmitting SIP request (444 bytes) to UDP:192.168.0.158:5060 --->
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.9:5060;rport;branch=z9hG4bKPj69db6ce7-c623-49ac-b8bf-84a18741ceb9
From: <sip:[email protected]>;tag=ff7e15e5-d474-4fb3-a143-56d4bad629ee
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: 16b52af3-35a4-4e27-bfe8-1dfe3df9bb19
CSeq: 21607 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-16.0.40.7(20.5.2)
Content-Length:  0


[2024-01-26 16:35:31] VERBOSE[2802215] res_pjsip_logger.c: <--- Transmitting SIP request (444 bytes) to UDP:192.168.0.158:5060 --->
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.9:5060;rport;branch=z9hG4bKPj69db6ce7-c623-49ac-b8bf-84a18741ceb9
From: <sip:[email protected]>;tag=ff7e15e5-d474-4fb3-a143-56d4bad629ee
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: 16b52af3-35a4-4e27-bfe8-1dfe3df9bb19
CSeq: 21607 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-16.0.40.7(20.5.2)
Content-Length:  0


[2024-01-26 16:35:35] VERBOSE[2802215] res_pjsip_logger.c: <--- Transmitting SIP request (444 bytes) to UDP:192.168.0.158:5060 --->
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.9:5060;rport;branch=z9hG4bKPj69db6ce7-c623-49ac-b8bf-84a18741ceb9
From: <sip:[email protected]>;tag=ff7e15e5-d474-4fb3-a143-56d4bad629ee
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: 16b52af3-35a4-4e27-bfe8-1dfe3df9bb19
CSeq: 21607 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-16.0.40.7(20.5.2)
Content-Length:  0


[2024-01-26 16:35:39] VERBOSE[2802215] res_pjsip_logger.c: <--- Transmitting SIP request (444 bytes) to UDP:192.168.0.158:5060 --->
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.9:5060;rport;branch=z9hG4bKPj69db6ce7-c623-49ac-b8bf-84a18741ceb9
From: <sip:[email protected]>;tag=ff7e15e5-d474-4fb3-a143-56d4bad629ee
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: 16b52af3-35a4-4e27-bfe8-1dfe3df9bb19
CSeq: 21607 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-16.0.40.7(20.5.2)
Content-Length:  0

[Stewart1 (Stewart)]

I know nothing about Janus but found SIP plugin documentation .
It says “This results in the plugin registering at the SIP server and acting as a SIP client on behalf of the web peer.”, i.e. the plugin registers to Asterisk.

But your ‘gotrunk’ trunk is attempting send registrations to 192.168.0.158 . Is this related to Janus? If so, please provide relevant documentation. If you are too new to post links, post them as preformatted text.
In any case, Asterisk is getting no response to both REGISTER and OPTIONS sent to 192.168.0.158 UDP port 5060. What should be there? What is the networking path between Asterisk on 192.168.10.9 and 192.168.0.158, which I assume are different subnets?