How do I find the differences in SIP extension settings?

I am trying to troubleshoot why some extensions appear to be not allowed to use dial the PAGE ALL number. They can receive page but they cannot dial the page number (“call failed”)

I wonder if the responsible setting is in FreePBX or in the settings of the IP-Phones…

(I am using RasPBX, FreePBX, Grandstream GXP1405)

Hi @newbie745 Can you provide the asterisk cli logs for the failed call to debug this.

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Thank you, I believed I have identified the problem just by looking at Reports > CDR Reports.
The extension that dialed the page via speakerphone, was never hung up, so it was busy and calls to it failed.
I am relieved that there was nothing more complicated.

I would like to find how to set a max time limit, so that such forgotten (walk-away) cases will self-resolve after say 15 minutes.

Is there a place to set a different max duration for PAGE calls, INTERCOM calls, and trunk calls?

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