Thanks,
I had to run the disc in rescue mode and run fdisk. This allowed me to
create a new partition table in MBR/Legacy mode. This resolved the issue.
It was a 500GB disk that previously had FreeBSD on it.
On Sun, Sep 30, 2012 at 8:31 AM, m...@tdiehl.org wrote:
On Sat, 29 Sep 2012,
CentOS has a known bug in it's installer that will not allow the install
disc to be run in UEFI mode. They had to re-release their x86_64 version
of 6.3 to resolve the issue. It's on their website. I had figured the
latest CentOS - 6.3 - was being used instead of 5.8.
On Mon, Oct 1, 2012 at
The issue was that the disk was already in GPT mode. I had to convert it
to MBR/Legacy mode.
On Mon, Oct 1, 2012 at 4:12 AM, Wrinkled Cheese wrinkledche...@gmail.comwrote:
CentOS has a known bug in it's installer that will not allow the install
disc to be run in UEFI mode. They had to
Hello guys,
my name is Gianluca and this is my first post in this ml.
i've a strange problem with my asterisk box. I'll try to explain you.
A (sip from ser) calls -- B (sip asterisk peer)
B put A on hold with musiconhold
B calls C
B transfer the call with A to C
A hears the C voice while C
Today I called some support hotline, for this support hotline no CDR was
created, also the call wasn't recorded, though there's a MixMonitor in my
dialplan, automatically recording every call.
Out of curiosity I set core set verbose 10 in the asterisk console. I
then dialed the support hotline
Did you take a look at the asterisk log? With core set verbose 3 or more?
Am 01.10.2012 12:46, schrieb Gianluca Baù:
Hello guys,
my name is Gianluca and this is my first post in this ml.
i've a strange problem with my asterisk box. I'll try to explain you.
A (sip from ser) calls -- B (sip
Hi,
The call waiting tone is very annoying (you hear nothing while it plays
the beep). I need callwaiting because of the queues (the phone has to
ring as soon as you hangup) but I want to remove the beep on my dahdi
channels, how can I do?
Thanks,
Niccolò
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http://www.linuxsystems.it
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-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Niccolò Belli
Sent: Monday, October 01, 2012 10:04 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How to remove the call waiting tone without
Il 01/10/2012 17:12, Danny Nicholas ha scritto:
I would start here
http://www.voip-info.org/wiki/view/Asterisk+indications+default
You could change the tone to something less annoying or just inaudible.
Does it affect dahdi channels? If I recall correctly the dahdi tones are
Maybe /etc/asterisk/chan_dahdi.conf
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Niccolò Belli
Sent: Monday, October 01, 2012 10:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Il 01/10/2012 17:47, Danny Nicholas ha scritto:
Maybe /etc/asterisk/chan_dahdi.conf
No, the only option here is to enable/disable callwaiting.
Cheers,
Niccolò
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http://www.linuxsystems.it
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Is it hardcoded in zonedata.c, am I right?
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New to Asterisk? Join us for a live introductory webinar every Thurs:
That would be correct. You could always modify a country you aren't in for
testing purposes, keeping in mind that for each change you would do a make
install on dahdi to recompile the module, then do a service asterisk stop;
service dahdi restart then service asterisk start to test properly.
I modified the Italian zone in zonedata.c from
{ DAHDI_TONE_CALLWAIT, 425/400,0/100,425/250,0/100,425/150,0/14000 },
to
{ DAHDI_TONE_CALLWAIT, 0/14 },
but I can still hear the damn beep :(
I even rebooted the pc, suggestions?
Niccolò
--
http://www.linuxsystems.it
--
This is probably a dumb question, but your country/zone is set to it
(installs as us by default)?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Niccolò Belli
Sent: Monday, October 01, 2012 12:46 PM
To:
Il 01/10/2012 20:08, Danny Nicholas ha scritto:
This is probably a dumb question, but your country/zone is set to it
(installs as us by default)?
Obviously :)
Anyway I think I found where the fucking bastard is hardcoded:
chan_dahdi.c in asterisk :)
I will change
#define
I can't find a clear procedure to lower musicOnHold volume!
Any suggestions?
Hereis my music.conf file
[default]
mode=files
directory=moh
Thanks in advance!
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AFAIK, there is still not a MOH volume control. What I did was to take my
moh wav files and run them through sox like this
$ cp -iv macroform-cold_day.wav macroform-cold_day_orig.wav
$ sox -v 0.9 macroform-cold_day_orig.wav macroform-cold_day.wav
This produces a file 90 percent as loud.
Hello
You should be modify the volume in the file, there are several
software for that, like wavepad .
Regards
On Mon, Oct 1, 2012 at 2:52 PM, Danny Nicholas da...@debsinc.com wrote:
AFAIK, there is still not a MOH volume control. What I did was to take my
moh wav files and run them
A long shot but how about 'campon' a queue, available on most old phones
systems but not asterisk.
Well maybe will still apply
https://issues.asterisk.org/jira/browse/ASTERISK-460
-Original Message-
From: asterisk-users-boun...@lists.digium.com
I have the following in sip.conf for asterisk running on localhost.
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.255
However, there is still repeated ill-intentioned access in this form:
NOTICE[17550]: chan_sip.c:14847 handle_request_invite: Call from '' to
extension '700972595637212'
You do not have an exten = 700972595637212 in the context in extensions.conf
that SIP device is set to in sip.conf
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Smith
Sent: Monday, October 01, 2012
Hi!
I've been confronted with an interesting issue to resolve. The
issue is located here:
https://issues.asterisk.org/jira/browse/ASTERISK-20163
The issue involves case-sensitivity of channel and global variables
in the dialplan. Current behavior is as follows:
1) Variables created in the
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Michelson
Sent: Monday, October 01, 2012 4:15 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Case-sensitivity of Dialplan variables.
Hi!
I am using Meeting on 1.4.43 with a handfull of devices, like 10 to 20
in a Meetme.
I can tell a difference (as two of the devices are close to each
other) that they are
not fully in sync. One was slightly behind the other... Any way to get
them more in sync?
Is it the delay from starting
- Original Message -
From: Jerry Geis ge...@pagestation.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, October 1, 2012 5:01:43 PM
Subject: [asterisk-users] MeetMe
I am using Meeting on 1.4.43 with a handfull of
Can anyone tell me if it is possible to invert the signaling bits on a T1
channel?
I need to emulate PLAR signaling in asterisk. EM seems to be an exact
match if reversed.
I need idle bits and seized
Thank you
--
_
On Mon, 1 Oct 2012, Jerry Geis wrote:
I am using Meeting on 1.4.43 with a handfull of devices, like 10 to 20
in a Meetme.
I can tell a difference (as two of the devices are close to each
other) that they are not fully in sync.
You would have to measure how many ms they are 'out of sync' to
Am 01.10.2012 22:13, schrieb Eric Smith:
I have the following in sip.conf for asterisk running on localhost.
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.255
Really on localhost only, no peers other than from 127.0.0.1? Not sure
why deny/permit is not working, but you could do
Am 01.10.2012 20:43, schrieb motty.cruz:
I can't find a clear procedure to lower musicOnHold volume!
Any suggestions?
exten = 1234,n,Set(VOLUME(TX)=-3)
exten = 1234,n,MusicOnHold(default)
in your extensions.conf should do the trick. (play around with the -3 value)
--
Nothing happens at the same time, unless you're broadcasting information
over some transport that supports multicast sends. There's always going to
be some interspersing of transmissions, if for no other reason than each
participant's channel in the conference has to be serviced after the media
- Original Message -
From: Jerry Geis ge...@pagestation.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, October 1, 2012 8:01:06 PM
Subject: Re: [asterisk-users] MeetMe not fully in sync
Mathew,
Makes sense does it
On 10/1/2012 5:15 PM, Mark Michelson wrote:
(HASH would be evaluated properly but hash would not). My personal
opinion is that all variable evaluations should be case-sensitive.
+1
case insensitivity to accommodate carelessness is evil. much easier for
NoOp to tell us SIP_CODEC is unset,
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