Playback using remote URI not working

Hello, I’m facing a problem when I try to execute the dial plan described below.

exten => _X.,1,Verbose(Testing Started)
exten => _X.,n,Playback(
exten => _X.,n,Hangup()

The dial plan is returning warnings when it tries to execute the Playback method and the call does not hang up. Here are the logs returned by the Asterisk.

== Setting global variable 'SIPDOMAIN' to 'XXX.XXX.XXX.XXX'
-- Executing [123@test_dial_plan:1] Verbose("PJSIP/test_lele-00000020", "Testing Started") in new stack Testing Started
-- Executing [123@test_dial_plan:2] Playback("PJSIP/test_lele-00000020", "") in new stack
  > 0x7f6338038760 -- Strict RTP learning after remote address set to: XXX.XXX.XXX.XXX:8000
  > 0x7f6338038760 -- Strict RTP qualifying stream type: audio
  > 0x7f6338038760 -- Strict RTP switching source address to
[Mar  8 14:18:20] WARNING[10439][C-00000021]: file.c:779 ast_openstream_full: File does not exist in any format
[Mar  8 14:18:20] WARNING[10439][C-00000021]: file.c:1252 ast_streamfile: Unable to open (format (ulaw)): No such file or directory
[Mar  8 14:18:20] WARNING[10439][C-00000021]: app_playback.c:492 playback_exec: Playback failed on PJSIP/test_lele-00000020 for
-- Executing [123@test_dial_plan:3] Hangup("PJSIP/test_lele-00000020", "") in new stack
== Spawn extension (test_dial_plan, 123, 3) exited non-zero on 'PJSIP/test_lele-00000020'

I already tried the approaches below.
• Set the cache settings on the media server
• Use different file extensions like (ulaw, wav, gsm)
• Omit the file extensions.

I can’t find any documentation that allows playback of anything but a local file with the filetype omitted. Either the documentation is wrong, or this is not yet supported.

I was supposing it based on this article.±+URI+Media+Playback

But it’s really strange that the current docs about the Playback method on Asterisk 16.x do not talk about using the URI as a parameter.

So should I consider that Asterisk does not have support for it?

16 has support for it. You may want to increase debugging[1] to see if there are underlying reasons. There are also modules required to be loaded, so if you are doing selective module loading I’d suggest turning on autoload and testing.

[1] Collecting Debug Information - Asterisk Project - Asterisk Project Wiki

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