Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
So I have done through chan_console.c and searched for console_pct_lock() - every one - has a matching console_pvt_unlock() How is the deadlock occurring ? jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
>>> How do we get this working For the time being, go back to 18.14.0 Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
>
>
> 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.
>
Well this is a bummer
[Sep 8 08:45:28] WARNING[313684][C-0001]: chan_console.c:651
console_indicate: Don't know how to display condition 26 on Console/default
--- <("<) --- Hangup on Console --- (>")> ---
[Sep 8 08:45:28] ERROR[313683]: utils.c:1027 lock_info_destroy: Thread
'stream_monitor started at [ 390] chan_console.c start_stream()'
still has a lock! - 'pvt' (0x55e647a245e0) from 'stream_monitor' in
chan_console.c:281!
How do we get this working
Jerry
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
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.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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@smvoice-mediacontroller:1]
SoftHangup("SIP/devgeis_to_nuc11cdev2-", "ALSA/dummy") in new stack
-- Executing [public_address@smvoice-mediacontroller:2]
Goto("SIP/devgeis_to_nuc11cdev2-",
"smvoice-mediacontroller-public-address,s,1") in new stack
-- Goto (smvoice-mediacontroller-public-address,s,1)
-- Executing [s@smvoice-mediacontroller-public-address:1]
NoOp("SIP/devgeis_to_nuc11cdev2-", "JERRY") in new stack
-- Executing [s@smvoice-mediacontroller-public-address:2]
Playback("SIP/devgeis_to_nuc11cdev2-", "beep") in new stack
> 0x7feeec0086b0 -- Strict RTP switching to RTP target address
192.168.1.8:17526 as source
-- Playing 'beep.gsm' (language
'en')
-- Executing [s@smvoice-mediacontroller-public-address:3]
Dial("SIP/devgeis_to_nuc11cdev2-", "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-
-- Channel Console/default joined 'simple_bridge' basic-bridge
[Sep 8 08:07:10] WARNING[282457][C-0001]: chan_console.c:651
console_indicate: Don't know how to display condition 26 on Console/default
-- Channel SIP/devgeis_to_nuc11cdev2- joined 'simple_bridge'
basic-bridge
> 0x7feeec0086b0 -- Strict RTP learning complete - Locking on source
address 192.168.1.8:17526
-- Channel SIP/devgeis_to_nuc11cdev2- left 'simple_bridge'
basic-bridge
-- Channel Console/default left 'simple_bridge' basic-bridge
[Sep 8 08:07:17] WARNING[282457][C-0001]: 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-'
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
