Displaying 20 results from an estimated 1004 matches for "danny".
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lanny
2020 Mar 20
4
Listenurl is not in HTTPS
...443/different.ogg
Nevertheless the listen URL still points to http://relay2.dgnet.be:443/different.ogg ( without https then )
So the links provided/generated by IceCast are wrong .
What did I miss ?
Is it possible to force the listenurl value in the icecast.xml ?
Thanks in advance for your help,
Danny
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2008 Apr 21
3
Decoding multiple Speex frames
Hello,
Could you please explain or send an example for decoding a packet that contains multiple Speex frames.
According to the API document the speex_decode_int() decodes only one frame of speech.
Best regards,
Danny Peled
R&D Software Engineer
Intelligence Solutions
NICE Systems Ltd. ISRAEL
*+ (972) 9 769 7161
? + (972) 54 4 602840
*Danny.Peled at nice.com <mailto:*Danny.Peled at nice.com%20>
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2020 Mar 21
1
Listenurl is not in HTTPS
...s this truly an OGG audio file? Why does the server response give a Content-Type of "audio/mpeg"?
Indeed, this is a relay of an audio/mpeg Shoucast stream .
Is this a problem ? should I change something in the config to be more compliant with Icecast best practices
Thanks for your help,
Danny
From: webmaster at berean-biblechurch.org <webmaster at berean-biblechurch.org>
Sent: samedi 21 mars 2020 01:57
To: Icecast streaming server user discussions <icecast at xiph.org>
Cc: Danny Gheraille <danny at gheraille.be>
Subject: Re: [Icecast] Listenurl is not in HTTPS
Danny...
2011 Oct 18
1
nvfaxdetect in 10.0
...We are moving our 1.4 asterisk with DAHDI over to 10.0 with
SIP. Everything is going nicely except that I can't get NV_FAXDETECT to
compile properly into 10.0. Because of this, I will have to have my
receptionist manually transfer incoming faxes. Any suggestions?
Thanks in Advance
Danny Nicholas
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2002 Aug 06
7
Mirroring 2 IDE Drive Linux
...o a Blank screen with an "L" in the upper left hand corner.
Can anyone tell me the proper commands to run with rsync to mirror from primary to secondary and boot with the copied drive properly with Redhat 7.1 ?
If not what other solution would you recommend.
Thanks
Sysadmin@ocsonline.cmo
Danny
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2011 Mar 04
5
Loudness of recorded wav-audio
Hello,
I sent a wav-audio to Asterisk though SIP and ISDN channels and recorded it
in wav-audio at the Asterisk server. I found the loudness level of the
recorded audio was too high comparing with the orginal audio. How can I
ajust it, so that there will be no amplifier used for recording.
Thanks a lot.
best regards
Felix
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2010 Jun 04
4
Asterisk on Ubuntu
...and DAHDI does not start at system start...i have to make
"/etc/init.d/asterisk start" and "/etc/init.d/dahdi start" manually every
time i reboot the machine (my laptop for testing)
So, what should i do in order to solve this situation?
Thanks in advance
Regards
--
Saludos
Danny Dias
SkypeID: danny.dias1
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2009 Feb 05
3
impossible packet length ...
...mpossible packet length (0) from nfs server
sunfire:/dist
Feb 5 17:01:03 warhol-00 kernel: impossible packet length (1647208041) from
nfs server sunfire:/dist
in this case the server is running Freebsd-7.0-stable, but I also get it when
the server is a
netapp.
is there a connection?
thanks,
danny
2005 Dec 15
5
initialize called more than once?
...print "\n initialize \n"
end
def hello
print "\n hello \n"
end
end
I thought initialize will be called exactly once when the controller is
constructed and not when an action is executed.
Thanks for your help and sorry if duplicate,
-Danny
2009 Jan 01
21
[Bug 19356] New: DPMS does not turn off LCD backlight on NV34M [ GeForce FX Go5200]
http://bugs.freedesktop.org/show_bug.cgi?id=19356
Summary: DPMS does not turn off LCD backlight on NV34M [GeForce
FX Go5200]
Product: xorg
Version: unspecified
Platform: PowerPC
OS/Version: Linux (All)
Status: NEW
Severity: normal
Priority: medium
Component: Driver/nouveau
2008 Jul 06
3
[Bug 16624] New: invalid pointer past to RRChangeOutputProperty
...thp about that.
To fix the problem in nouveau, either the property should be set up as Bool, or
we need to do an explicit cast. I provide a patch for the latter (because I am
not sure if we want more than 2 states for this property in the future). Let me
know if the former is preferred.
Cheers,
danny
--
Configure bugmail: http://bugs.freedesktop.org/userprefs.cgi?tab=email
------- You are receiving this mail because: -------
You are the assignee for the bug.
2009 Mar 22
4
7.2-PRERELEASE/sunx2200/bge/msi broken
Hi,
between March 16 and now, bge on a Sun X2200 stopped working,
turning off msi (via hw..pci.enable_msi=0) got it working again.
I tried first replacing bge with an older version but that did not help.
please advice :-)
Danny
2009 Jan 28
1
asterisk-users Digest, Vol 54, Issue 94
> Date: Wed, 28 Jan 2009 13:11:19 -0600
> From: "Danny Nicholas" <danny at debsinc.com>
> Subject: Re: [asterisk-users] SIP Registrations broken on 1.4.22.1?
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> <asterisk-users at lists.digium.com>
> Message-ID: <D32AD473FC574B41AE6...
2008 Dec 20
13
[Bug 19203] New: X default modes no longer added
...to
800x600 or 640x480. The gpu or panel would do the scaling.
Recently this is no longer possible. The edid only contains the native 1024x768
resolution and the log no longer mentions checking the modes from the builtin
modepool.
The lower resolutions were quite useful for some games and videos.
danny
--
Configure bugmail: http://bugs.freedesktop.org/userprefs.cgi?tab=email
------- You are receiving this mail because: -------
You are the assignee for the bug.
2007 Jan 31
3
Hi Honies! I'm home!
Many of you may have seen the recent announcement about Danny Windham
coming on as the new CEO of Digium. This is one of the most exciting
things to happen to Digium and to Asterisk at large. When Danny comes on
board, I will be transitioning to the role of Chief Technical Officer
(retaining my position of chairman of the board of directors), providing...
2018 Dec 28
2
LLVM Coverage Viewer
...be viewed locally. The tool is distributed from npm here
https://www.npmjs.com/package/llvm-coverage-viewer . The code is hosted on
github here https://github.com/hooddanielc/llvm-coverage-viewer. For any
questions or concerns, please let me know or make an issue on github page.
Hope this helps,
Danny Hood
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2010 Sep 02
4
agi playback to execute say.conf settings
Hi all,
I am using asterisk-1.6.2.10. I changed say.conf script for customized
number reading.
In the extension.conf:
--------------------------
[number-to-voice]
exten => 8765,1,playback(num:344345,say)
exten => 8765,n,hangup
It executes corresponding say.conf script and produces good results for me.
but when I write it in agi does not working. Here is agi debug output from
asterisk.
2020 Mar 20
1
Listenurl is not in HTTPS
...in https on IceCast ( https://relay2.dgnet.be/ ) , except this value listenurl ( in Admin / Mountpoint) which in cannot change apparently.
So the .m3u file is created with an http:// link ( instead of https:// )
But the stream is available and working in https://relay2.dgnet.be:443/different.ogg
Danny
From: Icecast <icecast-bounces at xiph.org> On Behalf Of Damien GENESTE
Sent: Friday, March 20, 2020 3:13 PM
To: Icecast streaming server user discussions <icecast at xiph.org>
Subject: Re: [Icecast] Listenurl is not in HTTPS
Hello,
If you are using liquidsoap, check the URL in your...
2012 Sep 26
6
SIP Retransmitting REGISTER message
Hi,
I was trying to register a VoIP trunk in Asterisk , where its keep on
sending Register message to the server, where I am not getting any response
from server.
But whereas if i register in Xlite softphone the account is getting
registered.
I suspect it could be network related issue, but since in softphone it is
getting registered from the same network.
Any ideas to isolate things would be
2012 Oct 31
2
Asterisk and OpenLDAP
Hello guys,
i would like to implement authentication for my sip extension with an
openldap server.
Following this guide
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ExternalServices_id291590.html
i see a template named [sip] to map the information of sip peers into ldap.
But i'm not interested to create a template, i would only authenticate
sip extensions using username