IVR Message not playing

Can someone assist please. My IVR message is not playing when I phone in to the free PBX sip trunk silent not even ring . I know the IVR is answering as it starts to ring at the time out destination after awhile. Checked the message was recorded by selecting under the join message in the conferencing application and I can here the message play when I dial the conference code all good there. If I change the destination to extension ring group I hear ringing and the extension or extensions rings no problem. Any suggestions would be appreciated

If you connect to the asterisk CLI, you should be able to see the server try to play the file and why it fails. Please post the output of the CLI, and we’ll start there.

Thank you for the response as I am very very new at this I am not sure what command to put in to see what you are looking for appreciate the assistance

In Linux:

asterisk -Rvvvvvvvvvv

Then place the call and it will show you what it’s trying to do and why it fails (hopefully)

so I have the cli command box open ready to send a command I would like to know what I type
in there please

It’s in my above post.

Its telling me no such command thank you

is it a linux CLI or asterisk CLI?

If it’s an asterisk CLI, you just need to set verbose to 10:

core set verbose 10

Then place the call.

it is now saying core verbose is still 10 make call and nothing else shows thank you

That’s really strange…it should be showing you SOMETHING

Thank you for the help not sure now what the best thing is to do

If you’re not able to get anything from the asterisk CLI, you might want to consider posting any CLI input/output and/or screenshots to try and identify the issue.

When you place a call with verbose set to 10, you should get at least a full screen of text from within the asterisk CLI (when viewing from within Linux). You might be using the web tool which is insufficient for this purpose…

Hope I am doing it correct to recap I am in GUI and under Admin go to asterisk CLI
CLI command and at the end it has send command .It is within this block I typed the command as above
again thank you

Seemed to have some success now went in via root on the pc running free PBX and typed in asterisk -Rvvvvvvvvvv shows CLI made call and a whole lot of stuff has just come on the screen with lots of red warning not sure how I am going to post it ? thank you

Does the above photo help in trying to solve my problem again thank you

There is a clue in there, your phones have a g729 license so calls will “pass-through”, your server however does not, hence no audio for voicemail nor IVR’s nor MOH that are handled by the asterisk box.

I appreciate every ones help thank you. I did have endless problems setting up the sip trunk in the beginning and it all had to do with the g729 codec giving me a big head ace as mentioned very new with the free PBX.
Is it possible from what you are seeing how to resolve the problem please

Either buy and install g729 licenses from Digium or don’t use it.

Apologies little confused as I I am in the phase of testing at the moment I do not have any physical phones as such have ordered grand stream waiting so I am using a bra app on apple and two micro sip soft phones on the pc to test with did the recording via the apple and I am basically phoning via my pstn to the sip trunk as mentioned the greeting is not playing mot sure what digium and licenses mean