I’ve tried multiple variations of this page - One with the Authenticate ID being the same as the SIP User ID and currently trying it without it. I’ve tried both the UCP password and the secret. Used the hostname and the IP for it, the SIP Server.
I’ve tried changing the Local SIP Port and TEL URI.
Is there something I have done wrong? The setting for the extension are pretty much default. Doing a packet capture on the phone and looking at it in Wireshark, I couldn’t see any SIP packets.
I went looking through the logs in /var/logs/asterisk and found this in fail2ban
[2023-06-21 14:13:08] NOTICE[8053] chan_sip.c: Registration from '<sip:[email protected]>' failed for '219.18.32.187:42613' - Wrong password
218 is a pjsip extension, would it be saying chan_sip? And I’ve copied the password directly from the extension so I’m not sure how it could be the wrong password.
Because the phone is incorrectly attempting to register to the chan_sip bind port.
In Asterisk SIP Settings, chan_pjsip tab, look for a section probably called 0.0.0.0 (udp) .
Check the value of Port to Listen On .
In the phone settings, the value of SIP Server should be the host name or IP address, followed by a colon and the port number you saw for Port to Listen On. For example, sip.example.com.au:5160
Of course, use the actual host name or IP address of your PBX, and the actual value of Port to Listen On.
If you still have trouble, post what now appears in the Asterisk log when it attempts to register.
The Local SIP Port setting in the Grandstream is the port number at the Grandstream end. Except in unusual circumstances it can be anything, with no effect.
Make sure that SIP Server has sip.example.com.au:5164
and retest.