Displaying 20 results from an estimated 1000 matches similar to: "chan_oss.c:572 oss_write: Unable to set device to input mode error"
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
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
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 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
2004 Apr 07
1
chan_oss.c:461: error: too many arguments to function `ast_queue_frame'
I got this compiling the new cvs code ...
any idea ?
Tnx !
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-04/07/04-11:28:50\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\"
2018 Feb 15
2
chan_oss.c: Unable to register channel type 'OSS'
Hi list!
Currently I use Asterisk 1.8.30.0 on an OpenWRT-Switch.
Now I want to change to Asterisk 13.14.1 on a Banana PI (with
Armbian/Debian 9).
Well, I copied the configuration and changed what needed, so
basically, it works, at least with my tests.
But when Asterisk will be started, in the message log I get this error:
[Feb 15 08:40:15] ERROR[3971] chan_oss.c: Unable to register channel
2018 May 04
0
How to constraint instructions reordering from patterns?
Here is a last example to illustrate my concern.
The problem is about the lowering of node t13.
Initial selection DAG: BB#0 '_start:entry'
SelectionDAG has 44 nodes:
t11: i16 = Constant<0>
t0: ch = EntryToken
t3: ch = llvm.clp.set.rspa t0, TargetConstant:i16<392>, Constant:i32<64>
t5: ch = llvm.clp.set.rspb t3,
2005 Sep 13
1
disable chan_skinny and chan_oss
How do I disable chan_skinny and chan_oss?
I think chan_skinny is associated with Cisco hardware, since I don't
have any I don't need this channel.
I just want to get rid of those warning messages at start up.
--
#Joseph
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
2008 May 12
1
Crappy sound on Console (chan_oss)
Hi all,
on my debian box i configured chan_oss to work with /dev/audio device.
CLI console command and Dial(CONSOLE/dsp) work perfectly but i notice
2 problems:
1. audio is very low in volume, even if i set 100 the mixer volume
(via cmd line setmixer utility)
2. the sound is very crappy: the voice is "vibrant", words sounds like
'ttthhhiiisss iiisss aaa ttteeessstt".
Seems
2018 Feb 15
2
chan_oss.c: Unable to register channel type 'OSS'
Zitat von Tzafrir Cohen <tzafrir.cohen at xorcom.com>:
Hi,
> Off-topic: any reason you don't use chan_alsa?
This was the "Armbian installation", I didn't configured it extra...
> Are you sure you quote the error message right?
Copy+Paste... ;)
But I searched a little bit and I really don't think, I need this module...
As I undestand, I just need it, if I
2004 Jun 18
0
not getting sound from chan_oss paging setup
Hi,
I am trying to setup an overhead paging system with asterisk. I have
followed some of the advice from the list and have oss.conf set for
autoanswer. The sound card and speakers work because they can play mp3s
just fine. When I call the extension, the asterisk console looks like
everything is working, but I get no sound. Here is what I get on the
console:
-- Executing
2018 May 04
0
How to constraint instructions reordering from patterns?
Krzysztof,
Thanks for your interest to my questions.
In order to clarify the context, here is the C source file of my test case.
The 3 builtins initialize some stack pointers. They have to be executed before any other instruction.
extern float fdivfaddfmul_a(float a, float b, float c, float d);
volatile static float x1,x2,x3,x4;
void _start(void)
{
float res;
2018 May 04
2
How to constraint instructions reordering from patterns?
The DAG dumping will try to print some of the nodes "inline" (i.e. where
they are used) to make the output more readable, so the dump of the DAG
may not strictly reflect the node ordering.
-Krzysztof
On 5/4/2018 8:18 AM, Dominique Torette via llvm-dev wrote:
> Here is a last example to illustrate my concern.
>
> The problem is about the lowering of node t13.
>
>
2014 Dec 12
1
Corrupt MixMonitor recordings - .gsm format
Hi all
Asterisk 1.8.11.0 on Centos 6.5
My VOIP phones are using G729 to a G729 trunk from a vendor (Centracom,
South Africa). Unlicensed G729 codec version on server.
75% of my .gsm files from MixMonitor are coming up corrupted about 3 minutes
into the recording.
The server has been up for 7 months beforehand with no problems with
recordings to .gsm format files.
I noted
2003 Oct 31
1
Problems with SIP
I'm new to Asterisk, but, Managed to get it working for outound calls from
my ATA --> Asterisk --> Cisco 2620 using SIP. However, I'm having problems
with Inbound calls from the Cisco.. Cisco 2620 --> Asterisk --> ATA .. In
fact, voice mail won't even work..
This is a snippet of what I'm getting when I try to call the ATA
-- Executing
2003 Dec 15
2
Beginners Question
Hi all,
New user to asterisk having just got it compiled and installed.
Running with no digium hardware (yet) and no soundcard in asterisk box.
Problem is using the sample configs with a sip phone added as follows
[2203]
type=friend
username=2203
secret=2203
host=dynamic
defaultip=192.168.0.2
dtmfmode=inband
canreinvite=yes
the console on * when running with -vvvvc says :- (whenb trying to
2004 May 21
0
unable to use EXEC in AGI
dear list
if I use EXEC in an agi script I get the following doing EXEC VoiceMailMain
-- AGI Script Executing Application: (VoiceMailMain) Options: ((null))
May 21 04:25:10 WARNING[1209214400]: chan_phone.c:422 phone_read: Error
reading:
Resource temporarily unavailable
May 21 04:25:10 WARNING[1209214400]: res_adsi.c:205
__adsi_transmit_messages: Un
able to send CAS
May 21 04:25:10
2004 Oct 05
0
Just getting started with Asterisk
Hi list,
I have been looking around for a while now, and cant seem to get to the
bottom of my problem.
My setup is that I have a separate SIP server that servers my SIP
subscribers, and I want to use Asterisk purely for voicemail for now.
So I set up a common SIP extension at my SIP server, and made Asterisk
register it, so that normal users can forward calls to that common
extension, and
2005 Jul 03
0
no sound. "Failed to write frame"
Hi all,
Couldn't find a place to search the list archives...
I'm having issues in getting any sound using a fresh asterisk install
and a SJPhone to connect to it. I went by the instructions pointed at
the "10 minute guide", located here:
http://www.voip-info.org/tiki-index.php?page=Asterisk+quickstart.
I installed it on a Slackware 10.1, by using no more than "make