[SOLVED] SIP registration failed on Doorphone Akuvox R20A

Hey everyone,

I’m a total noob on FreePBX and just got it running as a VM on Proxmox on my local network.

I first created two extentions via Application > Extensions > Add Extensions > Add new SIP Extension
On Linephone on my computer and smartphone everything worked well using the assistant and I was able to do a call between the two devices.

Created a third account to use with my Doorphone, which is an Akuvox R20A.

This is what I put in there and resulted in " Registration Failed"

Anybody got a clue?

Please provide the logs from FreePBX, see Providing Great Debug - Support Services - Documentation

Asterisk Logs by using tail -f /var/log/asterisk/full gives me

[2022-09-07 18:50:26] VERBOSE[28228] res_pjsip/pjsip_configuration.c: Endpoint 106 is now Reachable
[2022-09-07 18:50:26] VERBOSE[28228] res_pjsip/pjsip_options.c: Contact 106/sip:[email protected]:40582 is now Reachable.  RTT: 97.891 msec
[2022-09-07 18:55:38] VERBOSE[28228] res_pjsip_registrar.c: Added contact 'sip:[email protected]:54266' to AOR '105' with expiration of 3600 seconds
[2022-09-07 18:55:38] VERBOSE[28228] res_pjsip_registrar.c: Removed contact 'sip:[email protected]:55717' from AOR '105' due to remove existing
[2022-09-07 18:55:38] VERBOSE[28228] res_pjsip/pjsip_options.c: Contact 105/sip:[email protected]:55717 has been deleted
[2022-09-07 18:55:41] VERBOSE[28228] res_pjsip/pjsip_options.c: Contact 105/sip:[email protected]:54266 is now Unreachable.  RTT: 0.000 msec
[2022-09-07 19:00:38] VERBOSE[28228] res_pjsip/pjsip_configuration.c: Endpoint 105 is now Reachable
[2022-09-07 19:00:38] VERBOSE[28228] res_pjsip/pjsip_options.c: Contact 105/sip:[email protected]:54266 is now Reachable.  RTT: 45.018 msec
[2022-09-07 19:03:29] VERBOSE[11213] res_pjsip/pjsip_configuration.c: Endpoint 106 is now Unreachable
[2022-09-07 19:03:29] VERBOSE[11213] res_pjsip/pjsip_options.c: Contact 106/sip:[email protected]:40582 is now Unreachable.  RTT: 0.000 msec

FreePBX Debug with amportal a dbug is not working and gives me
-bash: amportal: Kommando nicht gefunden.

You also need to issue the CLI command “pjsp set logger on”.

Is 192.168.178.81 the router? If it is, you need to disable SIP ALG in the router.

If it is the doorphone, either its registration refresh logic is broken, causing it to refresh with changing port numbers, or Asterisk’s reply is getting lost, and it is periodically retrying the registration. However, this is unlikely if you have authentication enabled, as replies have to get through for the authentication stage to work.

Note that your log is showing the registrations succeeding on Asterisk, but with the registered address changing on each refresh.

Humm how about udp port settings??
Default SIP bind port use 5160.

I can’t find that option in https://www.j-ts.com/data/R20-manual.pdf and there is no problem with the bind port at the Asterisk end.

There is no “104” on /var/log/asterisk/full

My router (Fritzbox) has a little SIP server included that I’am using with the doorphone with no problems on Port 5160. That account is configured as Account 1 in the doorphone. Account 2 is freePBX.

Last night I deleted some of the extentions and created a new one “110” to try some more. Just fired up tail -f /var/log/asterisk/full again after booting the VM and getting this log now

[2022-09-08 09:56:22] VERBOSE[2288] res_pjsip/pjsip_options.c: Contact 105/sip:[email protected]:51048 has been deleted
[2022-09-08 09:56:25] NOTICE[2287] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"110" <sip:[email protected]>' failed for '192.168.178.45:5063' (callid: [email protected]) - Failed to authenticate
[2022-09-08 09:56:25] NOTICE[2287] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"110" <sip:[email protected]>' failed for '192.168.178.45:5063' (callid: [email protected]) - Failed to authenticate
[2022-09-08 09:56:25] NOTICE[2287] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"110" <sip:[email protected]>' failed for '192.168.178.45:5063' (callid: [email protected]) - Failed to authenticate
[2022-09-08 09:56:31] NOTICE[2287] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"110" <sip:[email protected]>' failed for '192.168.178.45:5063' (callid: [email protected]) - Failed to authenticate
[2022-09-08 09:56:31] NOTICE[2287] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"110" <sip:[email protected]>' failed for '192.168.178.45:5063' (callid: [email protected]) - Failed to authenticate
[2022-09-08 09:56:31] NOTICE[2287] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"110" <sip:[email protected]>' failed for '192.168.178.45:5063' (callid: [email protected]) - Failed to authenticate
[2022-09-08 09:56:56] VERBOSE[2287] res_pjsip_registrar.c: Added contact 'sip:[email protected]:61020' to AOR '105' with expiration of 3600 seconds
[2022-09-08 09:56:57] VERBOSE[2287] res_pjsip/pjsip_configuration.c: Endpoint 105 is now Reachable
[2022-09-08 09:56:57] VERBOSE[2287] res_pjsip/pjsip_options.c: Contact 105/sip:[email protected]:61020 is now Reachable.  RTT: 763.311 msec

So now the doorphones “110” is in the logs. 192.168.178.45 is the doorphones IP. My router is 192.168.178.1 by the way

The simplest cause of this is a wrong password.

Don’t ask my why and how but I was about to change the password in the doorphone, had already changed in in freePBX but not applied and then noticed that the doorphone was registered.

So now my doorphone is indeed registered. However, when I try to call the doorphone with *110 it’s not working and the log says

NOTICE[2300] res_pjsip_session.c:  105: Call (UDP:192.168.178.81:64762) to extension '*110' rejected because extension not found in context 'from-internal'.

Apparently some missconfiguration in an asterisk config file. How and what would I edit in freePBX?

why *110?? Should be 110.

Man, that’s it. Don’t know why I thought I have to add an * before
Thanks!

So now my doorphone calls my deviced and I can call the door phone and video is working as well
Solved thanks to you guys!

Your welcome.

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