Displaying 20 results from an estimated 10000 matches similar to: "Paging and conferences/chan_alsa."
2023 Sep 08
1
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
2006 Jan 17
2
MeetMe Listen Only flag (|m)
One of the features that I thought would be popular with the Web-MeetMe
suite is the ability to start all non-admin callers in a muted state and
selectively unmute them. For example any large conference that is
of an announcment nature with a Q&A session.
It's probably a feature I should have tested better, but I just
discovered
that a caller that is joined to a MeetMe with the |m flag
2005 Mar 21
1
iLBC codec and mute issues
I tried using the iLBC codec, and whlie I like it, I ran into a
strange issue. I did a few searches on Google and haven't found anyone
with the same issue as this.
Anyhow, I was using a Plantronics analog headset and box plugged into
a Digium TDM card, dialed out over my VoIP provider's IAX channel to
the PSTN.
I was in a conference call which is running on an Avaya PBX (which
2014 Mar 18
1
Which is more efficient for 1 to many broadcasting?
Putting a whole bunch of people into a listen-only/muted Confbridge
conference or getting the broadcaster audio into a MOH class and then
just having callers attach to that MOH class?
Does the the muted side of a Confbridge Room still try to mix in audio
from the muted channels or does it just disregard those channels and
only run mixes against unmuted channels?
Now, if the answer is
2023 Sep 07
2
Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
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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2023 Sep 08
1
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.
2023 Sep 13
2
Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
>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 not doing the hangup or the system.
jerry
--------------
2013 May 07
1
chan_alsa and confbridge
OK, somebody may have a much better way of doing what I'm attempting. If
so, I'm open to suggestions.
I am trying to configure confbridge to create a "conference" room with an
audio stream coming from my sound card. The idea is for a group of people
to be able to call in and listen to someone giving a speech but not
necessarily interact. I've got confbridge configured and
2003 Jul 22
0
IAX / MeetMe problem
Greetings,
I have a somewhat unique (I think) configuration that I am testing
involving MeetMe conferencing and have encountered a problem that I'm not
quite sure how to solve. Here is a brief description of my setup for the
background.
I wanted to offer the ability for users to mute and unmute themselves while
in a conference. If they enter a conference as monitor only, they are
2009 Oct 04
3
After call into console/dsp hangup hear ringing
I am running asterisk 1.4.26.1 and using ALSA not oss
dahdi 2.2.0
and libpri-1.4.10
I am calling into console/dsp I hear the audio just fine then after the
hangup I hear ringing
on the console/dsp.
Why would that be?
I found this bug for OSS https://issues.asterisk.org/view.php?id=13686
Does the same thing exist in ALSA???
some traces below
Jerry
== Parsing
2011 Oct 20
0
problems getting chan_alsa.so to run
Hi!
I am interisted to dial out from the console with chan_alsa. Can
somebody of you help me according this problem?!
I added user the asterisk to "pulse" and "pulse-access", and it didn't
change anything. alsa applications are routed by default to pulse.
cat /etc/asound.conf
pcm.!default {
type pulse
}
ctl.!default {
type pulse
}
What might be the problem?!
2007 Jul 15
2
1.4.7 chan_alsa : snd_pcm_open failed
asterisk-1.4.7, Fedora 7, intel emt64 - nocona:
== Parsing '/etc/asterisk/alsa.conf': Found
ALSA lib pcm_dsnoop.c:558:(snd_pcm_dsnoop_open) unable to
open slave
[Jul 15 10:12:23] ERROR[24420]: chan_alsa.c:365
alsa_card_init: snd_pcm_open failed: No such file or directory
[Jul 15 10:12:23] ERROR[24420]: chan_alsa.c:481
soundcard_init: Problem opening alsa I/O devices
== No sound
2007 Aug 05
0
chan_alsa - no sound / strange sound - 1.4.9
Hi
some problem with chan_alsa. Depending on the configuration I don't
get any sound output (output_device not set in alsa.conf - same as
output_device=default) or very strange output (output_device=hw:0,0)
when dialing into something like
exten => 10,1,Answer
exten => 10,n,Playback(soundfile)
exten => 10,n,Hangup
Other alsa applictions do work without problems and for example this
2015 Feb 16
0
LAN sip-to-sip
It looks as if that is more of a question/issue with your router, rather than Asterisk.
I have SIP devices working on my LAN, all hardwired, and have no need to open any ports or have the router address SIP in any way
My switch is not managed, and the router ports on the LAN side are all unmanaged, just a huge Ethernet "wirenut"
You SHOULD be able to communicate between devices on the
2005 Oct 02
0
Console Sound: Cuts out, Comes back after restart
I'm having a problem with sound output to the console.
My basic dial plan is as follows:
exten =>
_1NXXNXXXXXX,1,Dial(IAX2/####@voxee/${EXTEN},30,A(beep))
exten => _1NXXNXXXXXX,2,Playtones(info)
exten => _1NXXNXXXXXX,3,Hangup
I get the following output in the console:
___*CLI> dial 1#######@voxee
-- Executing Dial("ALSA/default",
2005 Sep 19
0
chan_alsa.c blocking sound port - solution
If anyone else is trying to use asterisk with the sound port AND use
something
else like mplayer my experience was asterisk BLOCKS the port.
I added a bug item this morning to suggest a parameter control in alsa.conf
and 1 line program change to chan_alsa.c of:
snd_pcm_nonblock(handle, 1);
Note this will always set NONBLOCK which is what I want at this time.
The paramter in alsa.conf is more
2003 May 27
0
Kernel Version for CAPI AVM Fritz PCI V2 /chan_capi /chan_alsa update to latest version
Hello there
I have a serious issue with the AVM Fritz PCI V2
I have the following setup and the problem is, that the kernel freezes hard after
about 16 hours. The second problem is, that the S-Bus gets jammed as well, so
you can't even use a analog phone! on the NT
Kernel 2.4.21rc2 with ACPI Patch and of course capi
are there any reasons why this configuration should not work?
And the
2003 May 27
1
Kernel Version for CAPI AVM Fritz PCI V2 / chan_capi / chan_alsa update to latest version..
Hello there
I have a serious issue with the AVM Fritz PCI V2
I have the following setup and the problem is, that the kernel freezes hard after
about 16 hours. The second problem is, that the S-Bus gets jammed as well, so
you can't even use a analog phone! on the NT
Kernel 2.4.21rc2 with ACPI Patch and of course capi
are there any reasons why this configuration should not work?
And the
2011 Sep 02
0
No subject
built-in; This doesn=92t matter because the moderator would have to use
meetmeadmin or the confbridge equivalent to control the other functions.
The moderator would either need two phones or a phone and a web =
interface.
Let=92s say Yves=92 =93special conference=94 is 5555. The moderator =
would start
using this command
Exten =3D> s,1,meetme(5555)
The participants would do
Exten =3D>
2004 May 23
1
*** Asterisk Sunday News: Conferences on the phone and IRL - "in real life"
Here in Sweden, it's supposed to be springtime. A wonderful time of the year,
with sunny skies and wonderful weather. Almost summer. Today, it's not.
It's winter all over again with rain and only 3 degrees celsius outside.
Better to stay inside and write a weekly Asterisk newsletter :-)
This week's topics:
-------------------
* Looking beyond Asterisk 1.0/1.1 - what's up?
*