Hello Jerry,
when you run asterisk using su, ownership of audio device files does not get
updated. When you login, you get the permissions. You can verify by ls -l and
getfacl on the device file.
Marek
--- Original Message ---
On Thursday, September 14th, 2023 at 14:33, Jerry Geis
On Wed, Sep 13, 2023 at 5:20 PM Jerry Geis wrote:
> >An issue[1] was already created by asterisk at phreaknet.org and they
> also put
> >a fix up for review and inclusion[2].
>
> >[1] https://github.com/asterisk/asterisk/issues/308
> >[2] https://github.com/asterisk/asterisk/pull/309
>
>
> The
>An issue[1] was already created by asterisk at phreaknet.org and they also
put
>a fix up for review and inclusion[2].
>[1] https://github.com/asterisk/asterisk/issues/308
>[2] https://github.com/asterisk/asterisk/pull/309
The change "seems" to be working.
Will test more tomorrow - had to
An issue[1] was already created by aster...@phreaknet.org and they also put
a fix up for review and inclusion[2].
[1] https://github.com/asterisk/asterisk/issues/308
[2] https://github.com/asterisk/asterisk/pull/309
On Wed, Sep 13, 2023 at 4:27 PM Jerry Geis wrote:
>
> I have found that I can
I have found that I can add the restart of asterisk (killall -9 asterisk)
to the h extension- BOY is that UGLY.
chan_console is not a testing device - how can we get this nasty bug fixed ?
Jerry
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> After a hung call, can you run core restart now from the asterisk console?
Doing a "killall -9 asterisk" is the only thing that works
I tried killall asterisk - does not free up the channel
the asterisk "core restart now" takes like a good 20 seconds to return but
does work.
The issue is I
It worked with my test. I'm on Asterisk 18.19.0
-- Executing [517xxx@voipms:4] System("IAX2/voipms-15815", "asterisk
-rx 'core restart now'") in new stack
-- Remote UNIX connection
Asterisk uncleanly ending (0).
Executing last minute cleanups
== Destroying musiconhold processes
>Using system() you could issue a asterisk -rx 'core restart now'
So I tried this
exten => s,1,Playback(beep)
exten => s,n,Dial(Console/default,20,g)
exten => s,n,Hangup
exten => s,n,System(asterisk -rx 'core restart now')
But it does not continue. Last thing I see is "Exited non zero"
so its
I have noticed that once my message speaks - the server thinks its done and
HUNGUP,
the endpoint STILL thinks the channel is active - the last message says
"Rx: BYE" on sip show channels
I tried ADDING to Dial() ,20,g and then had a Hangup after teh dial.
Its NOT getting there to hangup.
Jerry
--
>> Is there a dial plan call that can "exit asterisk" or completely reload
>> everything - killall active calls and start over ?
Using system() you could issue a asterisk -rx 'core restart now'
Doug
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Is there a dial plan call that can "exit asterisk" or completely reload
everything - killall active calls and start over ?
seems the console/dummy (chan_console) is holding some resource. How do I
just "exit" and start over after call came in ?
Thanks
Jerry
--
I don't know if this will help you, but looking back through an old config I
have for an older version of Asterisk, I had used chan_console with the old and
now defunct app_rpt app to listen to audio on various nodes via the console for
testing.
Here is what I did:
In console.conf, I defined
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
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>>> How do we get this working
For the time being, go back to 18.14.0
Doug
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Check out the new Asterisk community forum at:
>
>
> 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
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()
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
In the old days when I was using console/dsp, I would have to use
alsamix from the console to verify that the output wasn't muted. Maybe
it's still the same.
Doug
On 9/7/23 03:43 PM, Jerry Geis wrote:
ok switching to "Console/default" does show the text
--- <("<) --- Call to device
ok switching to "Console/default" does show the text
--- <("<) --- Call to device 'default' on console from 'default'
<2564286000> --- (>")> ---
--- <("<) --- Auto-answered --- (>")> ---
However I don't hear any audio.
Thanks
Jerry
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On Thu, Sep 7, 2023 at 3:27 PM Jerry Geis wrote:
>
> I found "console list available"
>
> ===
> === -
> === Device Name: default
> === ---> Default Input Device
> === ---> Default Output Device
> ===
I found "console list available"
===
=== -
=== Device Name: default
=== ---> Default Input Device
=== ---> Default Output Device
=== -
===
===
On Thu, Sep 7, 2023 at 3:20 PM Joshua C. Colp wrote:
> On Thu, Sep 7, 2023 at 3:15 PM Jerry Geis wrote:
>
>> Joshua
>>
>> Asterisk 18.14.0 with chan_alsa and Console/dsp works.
>> This does not work in 18.18.0 with chan_console enabled.
>> I am on Ubuntu 20.04 LTS.
>>
>> Is there a howto for
On Thu, Sep 7, 2023 at 3:15 PM Jerry Geis wrote:
> Joshua
>
> Asterisk 18.14.0 with chan_alsa and Console/dsp works.
> This does not work in 18.18.0 with chan_console enabled.
> I am on Ubuntu 20.04 LTS.
>
> Is there a howto for the new chan_console ?
>
I'm not aware of one. The module itself
Joshua
Asterisk 18.14.0 with chan_alsa and Console/dsp works.
This does not work in 18.18.0 with chan_console enabled.
I am on Ubuntu 20.04 LTS.
Is there a howto for the new chan_console ?
how can I get this working again ?
I am trying to just play audio on pulse audio.
Thanks,
Jerry
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