Displaying 20 results from an estimated 6000 matches similar to: "CONSOLE/dsp"
2005 Oct 15
1
No Audio from Console but mpg123 from shell worksfine.
Anyone have anything on this? (I'm sure someone will complain about me
bringing it up again, chill out...)
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jonathan
k. Creasy
Sent: Friday, October 14, 2005 10:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] No Audio
2009 Feb 09
1
chan_oss.c:585 setformat: Unable to re-open DSP device
== Manager 'sendcron' logged off from 127.0.0.1
vicidialnow*CLI> dial 919545090201
-- Executing AGI("OSS/dsp", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("OSS/dsp", "SIP/19545090201 at sip203||tTor") in new stack
-- Called 19545090201 at sip203
Feb 2 13:36:38
2005 Oct 16
1
No Audio from Console but mpg123 from shellworksfine.
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tzafrir
Cohen
Sent: Sunday, October 16, 2005 2:59 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] No Audio from Console but mpg123 from
shellworksfine.
>One possibility is that the volume is set to 0. aumix can be handy
here.
Does
2013 Aug 12
1
Asterisk 11.5.0 [Paging causes Asterisk to exit]
I've recently had an Asterisk 1.4.x install system crash, luckily I was well into configuring a replacement system.
Initially, the system was running:
Debian 6 64BIT
Asteirsk 11.3
USB sound card (to paging amp)
Audio file is sent to the console via /dev/dsp1 on a 3 second time delayed call file. Initial testing shows this working well, but the day before I was to deliver and install, I
2008 Mar 04
1
console dsp
I am trying to get a console/dsp application going with
1.4.18 and not hearing any audio. In the CLI I see the call coming in,
I see the Dial(Console/dsp)
I see <auto answered>
I see ALSA default
but I hear no audio.
What can I do to tell what is happening here.
I have in modules.conf:
noload chan_oss.so
load chan_alsa.so
For kicks I tried it the other way to noload chan_alsa.so and load
2005 Oct 14
0
No Audio from Console but mpg123 from shell works fine.
I get audio from mpg123 at the command line but when I load up asterisk
and try to get audio from the console it looks like it's working, and
even pauses like it is playing the file but there is no audio coming
from the speakers.
I have searched and looked through the archives and tried to fix this
but I have had no success. This is an onboard Intel card (AC'97) and I
also tried an SB
2007 Jan 09
0
Console\DSP
I am using a extension to dial the console which has autoanswer
enabled. I am getting a strange warning, has anyone seen this before?
Nothing on Google, or Voip-Info
[Jan 9 13:50:05] WARNING[5009]: chan_oss.c:1048 oss_request:
oss_request ty <console> data 0x0xb7851e00 <dsp>
<< Call to device 'dsp' dnid '(null)' rdnis '(null)' on console from
2007 Jul 27
4
CONSOLE=Phone/phone0 and CONSOLE=Console/dsp and Zap/g2
Hi List;
What the following mean:
CONSOLE=Phone/phone0
CONSOLE=Console/dsp
TRUNK=Zap/g2
I know SIP/John and Zap/1 but I do not know above (I
do not know also the difference between Zap/2 and
Zap/g2)?
Any advise?
Regards
Bilal
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2004 May 26
2
SPAM MESSAGE - [Asterisk-Dev] warning message (sound card) - when I run asterisk!!!
All,
After installing asterisk on Linux, I run "asterisk
-vvvc". But I got the following warning message:
chan_oss.so] => (OSS Console Channel Driver)
May 26 00:37:58 WARNING[-1084845952]: chan_oss.c:980
load_module: XXX I don't work right with non-full
duplex sound cards XXX
== Registered channel type 'Console' (OSS Console
Channel Driver)
== Parsing
2009 Dec 23
4
fax problem
Hello,
I need to send a tiff via fax with my asterisk 1.6.1.0.
I tried in the dialplan
[default]
exten => _X.,1,SendFax(/root/test.tiff)
but I have:
salledeconf1*CLI> console dial 111 at default
[Dec 23 16:24:22] WARNING[31739]: chan_oss.c:492 setformat: Unable to
re-open DSP device /dev/dsp: No such file or directory
-- Executing [111 at default:1]
2004 Nov 29
1
CONSOLE/dsp and command line play of wave file
Is there a way to have asterisk setup in ALSA mode (which is working)
but also be able to "play file.wav" and hear that also.
CONSOLE/dsp is working.
If I stop asterisk and "play file.wav" that works.
But if asterisk is running I cannot also play a file.wav
I am not actually doing them at the same time. Just a couple files
here and there to play. But as is I have to stop
2003 May 14
6
asterisk problem
the problem below keeps recarrying even after i have cleared this error when
i run asterisk -vvv or -c the error occurs again please help
..Warning, flexible rate not heavily tested!
.................WARNING[1024]: File loader.c, Line 212 (ast_load_resource):
/usr/local/lib/libh323_linux_x86_r.so.1: undefined symbol:
_ZN13PASN_Sequence17PreambleDecodeXERER11PXER_Stream
WARNING[1024]: File
2005 May 10
1
Group dial, first phone cannot pickup call. Cisco 7905 hangs.
I have a simple dial plan to cascade calls when the first phone does not
answer:
exten => 100,1,Dial(SIP/1000,10,tr)
exten => 100,2,Dial(SIP/1000&SIP/1001,10,tr)
exten => 100,3,Dial(SIP/1000&SIP/1001&SIP/1002,10,tr)
exten => 100,4,Voicemail(u100)
Problem is that the once the call goes onto the second and subsequent
steps exten 1000 cannot answer the call. When the user
2004 Jul 16
2
Offhook tone in channel OSS/dsp
Hi,
I have to develop a phone application using asterisk's
chan_oss.
When the phone is idle, i.e. the last command was a hangup,
one hears a "toot, toot, toot, ..."
But unforuntaly its use is in Germany, where one expects
a continous "toooooooooooooooooooooooooooooooooo ..."
before dialing.
Is there anything to define the tone indicating
"ready to dial"?
2003 Apr 17
5
X100P question
I have just started developing asterisk, and am trying to start simple. I
have a X100P device and an S100U device. I am trying to use the examples
provided, where I add a few lines to the /etc/zaptel.conf,
/etc/asterisk/zapata.conf, and /etc/asterisk/extensions.conf so that I may
connect an analog line to the X100P and an analog phone to the S100U. When
I dial the analog line, it should ring
2003 Nov 07
2
Callgroups and Pickupgroups in Console/dsp
Hi all.
I've made a patch for chan_oss.c to enable
callgroups and pickupgroups in it (since wasn't enabled).
I needed it for a special use of the console (pickup
calls arriving to the console from another phone)
btw, If someone is interested, I can submit a patch
to the bugtracker. I won't do it until
that's usefult for someone... since is a very special
features that probably no
2004 Dec 06
1
Console as extension problems
I'm trying to set up the console as an extension (so I can set up overhead
paging), but I can't seem to get it to work. When I call my paging extension,
I get an error that it can't open the device:
-- Executing Ringing("Zap/9-1", "") in new stack
-- Executing Dial("Zap/9-1", "Console/dsp0|18|A(new/whistle)") in new stack
<< Call
2004 May 02
1
phonejack and linejack in the same system
Hi,
I am a newbie in asterisk, i could compile it and run it with no problem on
a RedHat 9. In the same box, i got a linejack and a phonejack cards and i
downloaded the CVS driver from quicknet. This 2 card were working in a
openh323 (openphone and pstn) project with gnugk on a RedHat 9.
I am using the default samples, and i tried /dev/phone0 and /dev/phone1, but
when i run asterisk, i get this
2005 Jan 11
1
Dial Out Errors
Hey, I'm having some errors whenever I dial out and I can't dial in at
all. I'm using NuFone as my provider just so you know.
Jan 11 17:39:46 WARNING[1771]: chan_oss.c:413 soundcard_setinput:
Unable to re-open DSP device: No such device
Jan 11 17:39:46 WARNING[1771]: chan_oss.c:572 oss_write: Unable to set
device to input mode
Jan 11 17:39:46 WARNING[1771]: app_dial.c:359
2003 Sep 03
1
resend: * newbie: overhead paging and nbsd
I've rummaged through the archives and documentation and have yet to
find references to nbsd or mention of how to implement overhead paging
using chan_oss as mentioned in the list previously. I suspect that one
would use a soundcard in the PBX system and feed the output to speakers
and/or PA system. Would someone please point me to some procedures or
documentation to acomplish overhead paging?