Displaying 20 results from an estimated 200 matches similar to: "audio glitches in conference"
2010 Feb 10
1
problems with 1.6
In an attempt to fix problems with EAGI delays in 1.4 (see my other
message for more on that), I've tried upgrading to 1.6, in case it's a
bug that's fixed in the newer version.
Unfortunately, I'm having all kinds of trouble with this new install. My
system relies on conferences, and whenever I add any channel to it
(adding a SIP connection, playing an audio file, activating
2009 Aug 11
1
Weired Sound Issues on Opensuse 11.1 and ALSA (Wine 1.1.26)
Hello Everybody
I got some little issues with the sound. Logged in as my normal user - I've the testsound (winecfg) on the internal device. I can't change it.
If I log in as root I've the testsound on my external USB speakers (the way I would prefer it works).
I can't Imagine me - why there is a different between root and my normal user.
Please find below some details.
Code:
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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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
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
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
2010 Mar 01
1
Swift from eagi, problems with prosody rate
Hi, I'm trying to use Swift tts from eagi, my problem is when I send
EXEC SWIFT <*prosody rate*=\'.8\' >Hello World\, this is a test\,</*prosody*
>|0|1
Would I use a scape character?
Thanks
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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?
2003 Oct 12
1
AGI Test Fails
I've been trying to use the AGI get_data function for some time now, and
can't get it to work. Today I reinstalled a clean system with Red Hat
8.0 (I had been using RH9, but was told * had problems with RH9) and
downloaded the latest Asterisk CVS to install. I then downloaded and
installed perl-asterisk-0.08. I have extension 502 pointed at
EAGI(agi-test.agi). When I call that
2007 May 14
4
[*Win32 0.60] Sending call notification by e-mail/web?
Hello,
In case there are other users of the AsteriskWin32 port...
I haven't really used the AGI feature of Asterisk to run an application
from extensions.conf. *Win32 supports Perl, which I don't know. Apparently,
it's also possible to write AGI applications as EXE's (there's a
eagi-test.exe file installed by default).
=> When a call comes in, I'd like an AGI
2023 Jul 01
1
SetCallerPres command gone
The AGI debug command worked well, and I found the offending command:
SetCallerPres(allowed)
That worked in Asterisk 13, but from my google searching it looks like this command has disappeared in Asterisk 20 (actually everything after ver 13). I thought it was replaced with CALLERPRES(allowed) but this generated an error too in Asterisk 20.
Is there a replacement command?
-----Original
2006 Feb 08
2
sip channel status - how?
Hello!
I have an asterisk setup where several sip devices are connected to an
asterisk box. I am looking for a method that lets me know whether any of
the sip devices is on hook / off hook / busy etc.
I have tried the AGI command CHANNEL STATUS <channel name> but it
returns with a message
'There is no channel that matches <channel name>'
In concrete terms, my channel is
2003 Jun 18
1
Integration with external ASR engines
Hello,
Question for developers: what is the asterisk way to integrate with ASR
(speech recognition)?
Question to the community: has someone done anything in this direction?
On the first glance that shouldn't be too hard. One part is delivering audio
to the engine (for example,
main ASR players Nuance and Speechworks will be happy with RTP) - can be
done via RTP forking.
The other part is
2004 Apr 29
5
Start recording during call by pressing button sequence
Does anyone configure that or is that possible ?
Thanks in advance
--
Best regards
Vlad
2006 Apr 13
1
placing call with agi
I'm trying to set up a system so that I can record a conversation over
SIP. Monitor and the like don't work so well for me, because I need to
pipe the conversation to other programs in realtime, rather than record
to a file, so I've been trying to use EAGI instead. (if anyone has any
other suggestions about this, it would be greatly appreciated!)
At this point, I'm a little
2005 May 25
15
PHP/AGI Problem
Hi
I am currently developing a IVR application using
PHP/AGI. I am using the PHPAGI class at
http://phpagi.sourceforge.net/ to handle the
commuication with my *.
The application basically asks a caller to enter in
some information which is then processed and a answer
is read back out to them. I want the application to
loop back to the beginning after giving the answer so
they can try another
2003 Dec 29
1
Agent setup
Dear Group,
I have been successful in setting up the Agents, queues and getting agents
to log in.
Is there a way that I could configure the system so that the agent is called
back. i.e. the agent logs into the system, a call is destined for them and
their phone rings.
If some one has this setup I would be very interested in hearing from them.
Warm Regards and Thanks
---------------
Shad