Calls drop after 1 call (Internal and external)

[iammike]

Hi

I'm faced with the following problem that has given me the run around for the last few days and lead to several reinstalls of my 3cx instance.
I have had 3cx running in azure but want to bring it on prem as this is just for a home lab and testing for the production one I have in azure

Calling internally and externally I have the same issue.
The phone will start ringing caller side and the app will start ringing but then it will connect to voicemail,
User is showing present in the app and the management console.

Here is the log:

04/18/2021 6:17:09 PM - [CM503003]: Call(C:14): Call to <sip:[email protected]:0> has failed; Cause: 487 Request Terminated/INVITE from 127.0.0.1:5483

04/18/2021 6:16:29 PM - [CM503003]: Call(C:13): Call to <sip:[email protected]:0> has failed; Cause: 487 Request Terminated/INVITE from 127.0.0.1:5483

04/18/2021 6:13:48 PM - [CM503003]: Call(C:12): Call to <sip:[email protected]:0> has failed; Cause: 487 Request Terminated/INVITE from 127.0.0.1:5483

04/18/2021 6:11:03 PM - [CM503003]: Call(C:11): Call to <sip:[email protected]:0> has failed; Cause: 487 Request Terminated/INVITE from 127.0.0.1:5483

04/18/2021 6:09:33 PM - [CM503003]: Call(C:10): Call to <sip:[email protected]:0> has failed; Cause: 487 Request Terminated/INVITE from 127.0.0.1:5483

04/18/2021 6:09:23 PM - [CM503003]: Call(C:9): Call to <sip:[email protected]:0> has failed; Cause: 487 Request Terminated/INVITE from 127.0.0.1:5483

04/18/2021 6:06:43 PM - [CM503003]: Call(C:8): Call to <sip:[email protected]:0> has failed; Cause: 487 Request Terminated/INVITE from 127.0.0.1:5483

04/18/2021 6:03:58 PM - [CM503003]: Call(C:6): Call to <sip:[email protected]:0> has failed; Cause: 487 Request Terminated/INVITE from 127.0.0.1:5063

04/18/2021 6:02:37 PM - [CM503003]: Call(C:4): Call to <sip:[email protected]:0> has failed; Cause: 487 Request Terminated/INVITE from 127.0.0.1:5063

04/18/2021 1:16:16 PM - [CM503003]: Call(C:2): Call to <sip:[email protected]:0> has failed; Cause: 487 Request Terminated/INVITE from 127.0.0.1:5063

04/18/2021 1:13:26 PM - [CM503003]: Call(C:1): Call to <sip:[email protected]:0> has failed; Cause: 487 Request Terminated/INVITE from 127.0.0.1:5063

04/18/2021 1:09:11 PM - [EC100002] Connection with media server is lost

04/18/2021 1:08:54 PM - STUN discovery of public IP is disabled

04/18/2021 1:08:54 PM - Set log verbosity to MaxLevel = 2, Severity mask = 7

This instance is self hosted on hyper-v
Ports forwarded are :

I have also followed this guide:
https://www.slashadmin.co.uk/how-to-disable-sip-alg-on-unifi-usg/

I have also disabled "Disallow use of extension outside the LAN (Remote extensions using Direct SIP or STUN will be blocked)" for both extensions
The inbuilt firewall tester passes fine without issues

Any help is greatly apprecated

Thanks
Mike

 

[YiannisH_3CX]

Hi Mike,

The first thing you need to change is your audio ports. You need to forward all of them and not just the ports the firewall checker is checking. https://www.3cx.com/docs/ports/

Try calling *888 and see if that works from an extension. What this basically does is you make a call to the PBX, the PBX hangs up and calls you back. See if that happens and if you can hear your self. This shows you if the communication between your extension and the PBX is working properly.

Also you haven't clarified if your are calling between 2 extensions or if you are using a SIP trunk.