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 this:
[default]
input_device = default
output_device = default
autoanswer = no
context = <the context you want to use>
extension = <a valid extension>
callerid = <caller id>
language = en
overridecontext = no
mohintrepret = default
active = yes
In modules.conf I loaded the audio module (in this case it was chan_alsa.so,
but I also could use chan_oss.so). I made sure noload was commented out for
chan_alsa.so
In alsa.conf, I defined some of the same things as in console.conf:
[general]
autoanswer=no
context=<same as context in chan.conf>
extension=<same as extension in chan.conf>
inputdevice=plughw:0,1
otuputdevice=plughw:0,0
mute=true
You'll need to check your ALSA device to see what the input and output devices
are.
That last line is important, since on the console you may not have a mic that
works to talk, you just want to listen,
In extensions.conf, I defined a dialplan that instead of trying to dial out, it
just answered the call and then threw me into the app.
Then to dial from the console, I woudl use:
console dial <extension from console.conf>
And it woudl use the context I defined and launch the Rpt app.
What you could do is define somehting like this ,but have the extension use DISA so that
you can then get dumped into your normal dialplan logic where you could use "console
dial xxx".
No guarantees that this will work with a newer version of Asterisk, but this
did work with a 1.8 setup I used to have (that I have the configs saved for).
-Stacy
On 9/8/23 10:28 AM, Jerry Geis <jerry.g...@gmail.com> wrote:
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
