Displaying 20 results from an estimated 20000 matches similar to: "Monitoring calls via sound card"
2012 Mar 21
2
Echo cancellation with different sound card for speaker and microphone
I'm developing an application that have a video conference component.
For that I need echo cancellation, and is looking around for
algorithms/implementations of that, and the one in speex is an
alternative. In the documentation for speex I find the following
sentence however.
"Using a different soundcard to do the capture and plaback will *not*
work, regardless of what you may
2010 Feb 24
2
audio glitches in conference
I'm having a problem with conferences both meetme and app_conference,
though I've done most of the testing with meetme.
Essentially, I get little gaps in the audio - usually fewer than a dozen
or so samples, though it does vary. They seem to occur at random, but I
usually get one ever few seconds, on average. They also seem to delay
some buffer somewhere, so that if I start recording
2011 Feb 11
2
IceS output to alsa sound card
Hi.
I'm new to icecast.
I can see that IceS can source an input MIC, for example, from a sound card.
Can it also output a playlist to a sound card (to the speakers)?
If so, how?
Thanks
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2007 Apr 11
3
Execute EAGI script with params from extensions.conf
How can I execute an EAGI script with params from extensions.conf
Example python script:
InfMsg -s 1
in my extensions.conf
exten => 492,1,Answer
exten => 492,2,eagi,InfMsg -s 1
exten => 492,3,Hangup()
It doesn?t work
my * report...
-- Executing [92@telpin-112:2] EAGI("Zap/4-1", "InfMsg -s 1") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/InfMsg
2009 Feb 26
3
Getting SIP field P-Asserted-Identity from EAGI
Hi, using EAGI variables like
agi_request
agi_channel
agi_language
agi_type
agi_uniqueid
agi_callerid
agi_dnid
agi_rdnis
agi_context
agi_extension
agi_priority
agi_enhanced
agi_accountcode
I get a lot of data about a call, but I need to obtain P-Asserted-Identity
value from a SIP call. Are tehe any eagi variable to get that? Or have you
any solution??
Thanks!!!
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2019 Oct 08
2
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2003 Dec 18
11
Sphinx
Hi. I just started trying to play with Sphinx. I followed their site as far as running sphinx-server. It is listening on the default port. I copied sphinx2-simple to another file and changed sphinx2-continuous to sphinx2-server.
So, I ran eagi-sphinx-test under asterisk. What exactly is it supposted to do? Here's what I get:
debian:~# sphinx2-simple2
sphinx2-simple:
Demo CMU Sphinx2
2003 Oct 03
2
suggested hardware especially sound cards
Hello,
I've seen various suggestions thrown around for hardware when people ask,
but can we all agree on some basic hardware recommendations for a few basic
setups(and post them on a website) to make it easier for new people to avoid
some of the hardware/software pitfalls when they are setting up their first
systems.
Something like this:
(THIS IS JUST A PROPOSED LAYOUT SO PLEASE BE GENTLE)
2007 Aug 27
1
Detecting tones
Hello folks,
I'm interested in detecting tones on specific frequencies with
specific timing; for example, I'd like Asterisk to dial out and when
the channel starts/call connects, listen for a 1200Hz tone that plays
for 100ms.
Is this doable with Asterisk using something already extant? After
looking through documentation, mailing lists, and some of the source I
had the idea that I might
2003 Apr 11
1
Weird AGI/X100P behavior
I've got a single phone line coming into an X100P.
In extensions.conf I've got this:
[inboundzap]
exten => s,1,Answer
exten => s,2,EAgi,hanguptest.agi
I see the ring come in and Asterisk detects it and tries to do something
with it:
NOTICE[20492]: File chan_zap.c, Line 4017 (ss_thread): Got event 2
(Ring/Answered)...
-- Executing Answer("Zap/1-1", "") in
2010 Jun 11
1
WARNING message when play
When I use an eagi script when play a message appear a lot of warning
messages, but it play very well
I?m using
Asterisk 1.4.32
dahdi-linux-2.3.0.1
chan_ss7-1.4.1
Any ideas??
-- Playing 'ser002' (escape_digits=0123456789*#) (sample_offset 0)
[Jun 11 18:12:45] WARNING[15807]: file.c:1300 waitstream_core: write()
failed: Broken pipe
[Jun 11 18:12:45] WARNING[15807]: file.c:1300
2003 Dec 18
1
AGI and broken pipe
Hi All,
I was able to track down what I believe is a bug when using AGI
services. This bug may crash your system if your extensions.conf script
is intensive in using AGI services. Depending on your system's ulimit, *
keeps opening files until it reaches the system limit and then stops
responding.
Function app_agi/launch_script seems to leave an open and unused file.
Can someone confirm this?
2004 Sep 02
2
${CALLERID}
Hi,
need a quick help ... it should be easy but ...
exten =>_9898,1,Answer
exten =>_9898,2,VoiceMailMain(${CALLERID}@domain)
Accepting overlap call from '342' to '9' on channel 0/2, span 3
-- Executing Answer("Zap/8-1", "") in new stack
-- Executing VoiceMailMain("Zap/8-1", "@domain") in new stack
As you can see there
2013 Mar 20
2
xmpp priority setting and GoogleVoice
I just wanted to send out some information that will hopefully help
others. I don't know, maybe I'm the only one that's been having
problems with this. I've been pulling my hair out for a while
wondering why Google would not send my incoming calls to my Asterisk
box. The calls would just roll to voice mail and no packets ever
reached Asterisk. This has happened on two separate
2005 Mar 09
1
Paging using multiple sound cards/channels
Does anyone know if its possible to have more than one sound card in an Asterisk box and use each one as a paging zone? How about left and right channels of a single sound card? I'm looking to have 2 paging outputs if possible - I've read about using a Grandstream phone on autoanswer but I'd prefer to have the feeds come directly from the * box and go into a stereo amp feeding 2
2010 Feb 08
2
How to run a remote PHP script and still have access to audio stream?
Hello there,
I'm trying to figure out how to run a PHP script on a remote machine and
still have access to the audio stream associated with the call.
Ideally, I'd love to play/record audio files directly from/to the remote
server without having to copy them back and forth to the Asterisk
server. What is the best way to do this?
Is it possible to combine EAGI with FastAGI in PHP?
2013 Jul 09
1
Adjusting confbridge call quality
Is there any way I can improve the audio quality in a confbridge in
Asterisk 11? I've changed the internal_sample_rate setting to 44100
but that doesn't seem to make any difference. I would also think this
would make my confbridge recordings 44100 but they all end up as 8000.
Am I completely missing something?
--
Chris
2005 Jun 15
0
Sound card under CentOS 4
I am swapping my laptop over to CentOS and for some reason alsa swapped
my headphone jack and speakers designations during the install. I got
it so that my sounds work but was wondering how to swap them back so
they are defined/labeled right and why it might have done it in the
first place Thanks for your help
-----------------------------------------
Disclaimer: This electronic message,
2011 Feb 12
1
IceS output to alsa sound card
Hi Guys,
I'd reflect Vieri's question actually, because there must be a way of
doing it? Maybe not with ICES, but maybe combines with JackIT?
The reason for my interest is I work in a radio station in Spain, and
because of the geographics where I am we have to relay around a lot.
If there is a way of sending an incoming feed into ICES, and sending
it out in various directions, would
2004 Dec 06
2
SoftPhone on * with X-Lite or iaxComm (1 X100P card)
To all:
I am having an echo problem with X-Lite and iaxComm. I am using the
monitor speakers and a desktop microphone. My problem is that the sound
from the speakers is repeated by the microphone causing an echo that is
annoying. Is this correctable? I have searched throught this archives
and played with various settings but have not been able to fix this. I
have purchased a couple if