asterisk users - Sep 2023 - Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

Some progress to report:

I had to run asterisk as the user logged in -  actually not even that. I
could not "su user -c " to that user - I had to run it as that user.
Then I did a test and got audio! Great...
But when I do a second test. Asterisk HANGS on ChanIsAvail()

Then I thought lets SKIP that - and just let it do the Dial() - I stopped
everything - got it running again. - and then the Dial() hangs on the
second call.

So both ChanIsAvail() or Dial() both hang on the second call in.

So only 1 call in will work.
Below is the CLI report of the call that works.

This is my context
[smvoice-mediacontroller-public-address]
exten => s,1,ChanIsAvail(Console/default)
exten => s,n,GotoIf($["${AVAILCHAN}" =
""]?smvoice-busy,s,1)
exten => s,n,Playback(beep)
exten => s,n,Dial(Console/default)
exten => s,n,Hangup

Now what ???

Jerry


onnected to Asterisk 18.18.0 currently running on nuc11cdev2 (pid = 282195)
  == Using SIP RTP CoS mark 5
       > 0x7feeec0086b0 -- Strict RTP learning after remote address set to:
192.168.1.8:17526
    -- Executing [public_address at smvoice-mediacontroller:1]
SoftHangup("SIP/devgeis_to_nuc11cdev2-00000000",
"ALSA/dummy") in new stack
    -- Executing [public_address at smvoice-mediacontroller:2]
Goto("SIP/devgeis_to_nuc11cdev2-00000000",
"smvoice-mediacontroller-public-address,s,1") in new stack
    -- Goto (smvoice-mediacontroller-public-address,s,1)
    -- Executing [s at smvoice-mediacontroller-public-address:1]
NoOp("SIP/devgeis_to_nuc11cdev2-00000000", "JERRY") in new
stack
    -- Executing [s at smvoice-mediacontroller-public-address:2]
Playback("SIP/devgeis_to_nuc11cdev2-00000000", "beep") in
new stack
       > 0x7feeec0086b0 -- Strict RTP switching to RTP target address
192.168.1.8:17526 as source
    -- <SIP/devgeis_to_nuc11cdev2-00000000> Playing 'beep.gsm'
(language
'en')
    -- Executing [s at smvoice-mediacontroller-public-address:3]
Dial("SIP/devgeis_to_nuc11cdev2-00000000",
"Console/default") in new stack
  --- <("<) --- Call to device 'default' on console from
'MyName Here'
<2564286000> --- (>")> ---
  --- <("<) --- Auto-answered --- (>")> ---
    -- Called Console/default
    -- Console/default answered SIP/devgeis_to_nuc11cdev2-00000000
    -- Channel Console/default joined 'simple_bridge' basic-bridge
<e6f6e4e9-aa1f-452d-883d-65c4d93c59b1>
[Sep  8 08:07:10] WARNING[282457][C-00000001]: chan_console.c:651
console_indicate: Don't know how to display condition 26 on Console/default
    -- Channel SIP/devgeis_to_nuc11cdev2-00000000 joined 'simple_bridge'
basic-bridge <e6f6e4e9-aa1f-452d-883d-65c4d93c59b1>
       > 0x7feeec0086b0 -- Strict RTP learning complete - Locking on source
address 192.168.1.8:17526
    -- Channel SIP/devgeis_to_nuc11cdev2-00000000 left 'simple_bridge'
basic-bridge <e6f6e4e9-aa1f-452d-883d-65c4d93c59b1>
    -- Channel Console/default left 'simple_bridge' basic-bridge
<e6f6e4e9-aa1f-452d-883d-65c4d93c59b1>
[Sep  8 08:07:17] WARNING[282457][C-00000001]: chan_console.c:651
console_indicate: Don't know how to display condition 26 on Console/default
  --- <("<) --- Hangup on Console --- (>")> ---
  == Spawn extension (smvoice-mediacontroller-public-address, s, 3) exited
non-zero on 'SIP/devgeis_to_nuc11cdev2-00000000'
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On 9/8/2023 8:18 AM, Jerry Geis wrote:
> But when I do a second test. Asterisk HANGS on ChanIsAvail()
>
> Then I thought lets SKIP that - and just let it do the Dial() - I 
> stopped everything - got it running again. - and then the Dial() hangs 
> on the second call.
>
> So both ChanIsAvail() or Dial() both hang on the second call in.
>
> So only 1 call in will work.
> Below is the CLI report of the call that works.
>
> This is my context
> [smvoice-mediacontroller-public-address]
> exten => s,1,ChanIsAvail(Console/default)
> exten => s,n,GotoIf($["${AVAILCHAN}" =
""]?smvoice-busy,s,1)
> exten => s,n,Playback(beep)
> exten => s,n,Dial(Console/default)
> exten => s,n,Hangup
Not sure if this is the same thing you're seeing, but chan_console 
currently has a known deadlock issue that has not been resolved: 
https://issues-archive.asterisk.org/ASTERISK-30481
It's quite easy to reproduce, so it may be the case that the module is 
currently unusable.