It's here: http://queuemetrics.com/download/qloaderd-1.17.tar.gz
It's technically a part of QueueMetrics, but it does not require a licence
to run.
Feel free to use it. :)
l.
2009/8/18 Miguel Molina mmol...@millenium.com.co
Lenz Emilitri escribió:
You should log to a file and use a piece of
hello
is it possible to lock a conference IF no admin are connected ?
or how to do to have a conference offline?
thank you
Cordialement,
BERGANZ François
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
___
--
Nope - but you are also running on an unsupported version of asterisk,
so I am not surprised. From the readme:
===[ Installation Overview ]===
It is required that the proper version of Asterisk is installed prior to
installing Skype For Asterisk.
I was copying tracks from CD into mp3 files so that I could use it in
Asterisk 1.4.21.2 MOH. (BTW, I have already secured proper license to
play MOH to callers.)
I used MS Media Player version 11 and rip it at 128kbps (smallest) but
whenever I listen to MOH, I saw the following message on the
Oops sorry, the Asterisk version should read 1.4.26.1
On Wed, 19 Aug 2009, Julian Lyndon-Smith wrote:
Nope - but you are also running on an unsupported version of asterisk,
so I am not surprised. From the readme:
===[ Installation Overview ]===
Julian Lyndon-Smith wrote:
Nope - but you are also running on an unsupported version of asterisk,
so I am not surprised. From the readme:
===[ Installation Overview
]===
It is required that the proper version of Asterisk is installed prior
Asterisk Development Team wrote:
As posted on blogs.digium.com today:
http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/
the Asterisk project has changed providers for Music-On-Hold (MOH)
content distributed with/for Asterisk. In addition to the change for
future Asterisk
Please ignore my stupid reply to this, I was having issues with weasles
at the time.
___
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AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net
On Wed, Aug 19, 2009 at 09:28:24AM +0100, Thomas Kenyon wrote:
Great to hear, although I am a bit suspicious, the asterisk-sounds
package in
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ still has
a Mar 06 timestamp.
http://downloads.asterisk.org/pub/telephony/sounds/
has
hi,
i want CDR entry in database for a call which originated from manager via
action: originate
currently i didnt get this entry into my DB
any one have idea regarding this for getting this on DB
i enabled cdr_manager.conf entry to 'yes'
thanks in advance
regards
Dhaval
Since 2004 asterisk/libpri have been able to receive the Calling Sub Address
in the ISDN setup message, and the dialplan was able to use it if required.
It's support is limited to only NSAP, not BCD or user formatted.
At the time 25/06/04 the questioned was asked, wouldn't it be a good idea to
On Wed, 2009-08-19 at 09:16 +0200, BERGANZ François wrote:
hello
is it possible to lock a conference IF no admin are connected ?
or how to do to have a conference offline?
snip
If I understand you correctly, we are doing something similar. When
users call into a conference, they hear
Hello,
I need CAP_FOWNER=ep for the asterisk process, i set it with setcap on
the file /usr/sbin/asterisk, it's there when i look on it with getcap,
but after starting and loocking with getpcaps there's only
cap_net_admin+ep set.
So how exactly do I set CAP_FOWNER? Do I have to patch and
Daniel,
I'm a little confused as to what I'm seeing here. You're bounding
through two RFC1918 address networks -- 10.1.0.X and 192.168.2.X. Is
this some sort of dual NAT scenario?
Perhaps if you can explain a little more about your network setup.
N.
Daniel Bareiro wrote:
-BEGIN PGP
Hellos,
I have astersist 1.2 working with freepbx. I want to tie pin codes to
extensions(users). How do I do this?
--
Best Regards,
James Mutuku Ndeti
Agile Systems Limited
+254722490994
www.agile.co.ke
mutuku.wordpress.com
Has your organization implemented a customer relationship management
a few days ago slashdot (sorry i havent the link now) wrote about skype has
a very huge problem whit a licence in a core codec, and if they dont get an
aregment whit the codec owner they will close the doors...
David
2009/8/19 Thomas Kenyon dig...@sanguinarius.co.uk
Julian Lyndon-Smith wrote:
I wonder if that was not a codec specific issue, but rather the matter
of their license to the p2p technology provided by JoltID? Since Skype
has recently dveloped their own codec (SILK) they could easily drop
support for any codec that they previously licensed from outside. I
think that the
Michael Graves wrote:
I wonder if that was not a codec specific issue, but rather the matter
of their license to the p2p technology provided by JoltID? Since Skype
has recently dveloped their own codec (SILK) they could easily drop
support for any codec that they previously licensed from
Hi All,
This is my first post. I searched the archives and found something similar
and I tried some of those suggestions. I changed the file permissions on the
scripts directory to 777 (which doesn't seem secure), I also manually ran
the detectdahdi.sh script. The response is None.
I am running
On Wed, 19 Aug 2009, ABBAS SHAKEEL wrote:
Rsync looks really great !
Yep. Something I wish I learned about earlier in my career :)
How you are get CDR records etc from the remote servers for reporting
purpose . I was thinking to have one centralized database ? but your
comments let me
On Wed, 19 Aug 2009, Lee, John (Sydney) wrote:
I was copying tracks from CD into mp3 files so that I could use it in
Asterisk 1.4.21.2 MOH.
Are there any Asterisk+Audio expert that can offer me some advice?
Don't use MP3. Why would you want to burn CPU cycles decompressing the
same stuff
Boehm, Matthew wrote:
MeetMe requires an external timing source. Right now, using the
dummy
driver. Is it possible to use the card solely for timing purposes?
Any
benefit to doing so? Or should I just sell the cards?
DAHDI 2.2.0 provides timing without using the dummy driver and
I put a post on here about my issues with outbound calls not ringing but i
haven't resolved it. so i am trying again.
When i dial any outside number i dont get a ring tone at all. when the person
picks up and starts to talk i can hear them fine. it sounds great. How do I
start to troubleshot
Hi
I Have a problem with mysql/asterisk realtime interaction.
each time I try to connect a sip phone or to use this CLI command - realtime
mysql status - I obtain this error message :
Mysql Realtime: Failed to connect database server asteriskdb on localhost
(err 2002)
here a sample of my
Have you tried putting a (,r) on your Dial command (dial
dahdi/1/18005551212,60,r) ?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose
Sent: Wednesday, August 19, 2009 8:55 AM
To: asterisk-users@lists.digium.com
Steve Edwards wrote:
On Wed, 19 Aug 2009, Lee, John (Sydney) wrote:
I was copying tracks from CD into mp3 files so that I could use it in
Asterisk 1.4.21.2 MOH.
Are there any Asterisk+Audio expert that can offer me some advice?
Don't use MP3. Why would you want to burn CPU cycles
On 19 Aug 2009, at 14:59, harry R wrote:
Mysql Realtime: Failed to connect database server asteriskdb on
localhost (err 2002)
here a sample of my res_mysql.conf file :
[general]
dbhost = localhost
dbname = asteriskdb
dbuser = asterisk
dbpass = asterisk
dbport = 3306
mysql -uasterisk
Kevin P. Fleming kpflem...@digium.com writes:
DAHDI 2.2.0 provides timing without using the dummy driver and without
needing any cards. You can use a card for timing, but you'd need to hook
up a crossover cable between two of the ports to get them out of red
alarm status... but it's really
Have you posted a bug describing the issues you are having at
http://betareports.digium.com/mantis/
yet? I would love to have the opportunity to actually fix any bugs
that people find. :-)
I installed the 1.0 release of Skype for Asterisk and last night on my
production box running
Tzafrir Cohen wrote:
On Tue, Aug 18, 2009 at 10:00:55AM -0400, Dave Fullerton wrote:
Here's my $0.02. If you don't want an echo canceller, specify
echocanceller=none,x-y and have dahdi_cfg print a warning (at any
verbosity level) when an echo canceller is not specified for a channel.
On Wed, 19 Aug 2009, Dave Fullerton wrote:
Tzafrir Cohen wrote:
On Tue, Aug 18, 2009 at 10:00:55AM -0400, Dave Fullerton wrote:
Here's my $0.02. If you don't want an echo canceller, specify
echocanceller=none,x-y and have dahdi_cfg print a warning (at any
verbosity level) when an echo
On Wed, Aug 19, 2009 at 10:11 AM, Benny Amorsenbenny+use...@amorsen.dk wrote:
Kevin P. Fleming kpflem...@digium.com writes:
DAHDI 2.2.0 provides timing without using the dummy driver and without
needing any cards. You can use a card for timing, but you'd need to hook
up a crossover cable
Jeff LaCoursiere wrote:
On Wed, 19 Aug 2009, Dave Fullerton wrote:
Tzafrir Cohen wrote:
On Tue, Aug 18, 2009 at 10:00:55AM -0400, Dave Fullerton wrote:
Here's my $0.02. If you don't want an echo canceller, specify
echocanceller=none,x-y and have dahdi_cfg print a warning (at any
verbosity
There is another way,
Try to recompile your asterisk with this options:
1) edit asterisk Makefile and add:
BUSYDETECT+= -DBUSYDETECT_COMPARE_TONE_AND_SILENCE -DBUSYDETECT_MARTIN
2) Run
make clean
./configure --prefix=/usr
make
make install
Regards,
On Wed, Aug 19, 2009 at 6:26 PM, Luis
Dave Fullerton wrote:
It is true that this method would require more configuration work and
that it would probably throw people off who were used to the old method.
However, I don't agree that it leaves more room for error. The current
system, IMHO, has a certain amount of ambiguity to it.
Benny Amorsen wrote:
Kevin P. Fleming kpflem...@digium.com writes:
DAHDI 2.2.0 provides timing without using the dummy driver and without
needing any cards. You can use a card for timing, but you'd need to hook
up a crossover cable between two of the ports to get them out of red
alarm
On Wed, 2009-08-19 at 13:54 +, Ott Rose wrote:
I put a post on here about my issues with outbound calls not ringing
but i haven't resolved it. so i am trying again.
When i dial any outside number i dont get a ring tone at all. when the
person picks up and starts to talk i can hear them
Kevin P. Fleming wrote:
Dave Fullerton wrote:
It is true that this method would require more configuration work and
that it would probably throw people off who were used to the old method.
However, I don't agree that it leaves more room for error. The current
system, IMHO, has a certain
mysql -uasterisk -pasterisk asteriskdb
When I do that in a linux terminal it works.
But I always have this err 2002.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register
Dave Fullerton wrote:
The dahdi_scan tool will tell you whether hardware echocans are present
or not, among other methods.
I tried that, but I didn't see anything that specified whether the echo
canceller was present. Here's the output, can you tell me what I should
be looking for?
On Wednesday 19 August 2009 05:54:32 Raimund Sacherer wrote:
I need CAP_FOWNER=ep for the asterisk process, i set it with setcap on
the file /usr/sbin/asterisk, it's there when i look on it with getcap,
but after starting and loocking with getpcaps there's only
cap_net_admin+ep set.
So how
On Wed, 19 Aug 2009, harry R wrote:
mysql -uasterisk -pasterisk asteriskdb
When I do that in a linux terminal it works. But I always have this err
2002.
Going out on a shaky limb I know little about...
I always specify all of the connection options in scripts so I don't get
caught by
Kevin P. Fleming wrote:
Dave Fullerton wrote:
The dahdi_scan tool will tell you whether hardware echocans are present
or not, among other methods.
I tried that, but I didn't see anything that specified whether the echo
canceller was present. Here's the output, can you tell me what I
On 19 Aug 2009, at 16:37, harry R wrote:
mysql -uasterisk -pasterisk asteriskdb
When I do that in a linux terminal it works.
But I always have this err 2002.
I'd try and tcpdump it if you can find a way. Might be something odd
happening.
S
___
we are using Aastra 57i
i don't see that setting. where is it at?
From: jsulli...@opensourcedevel.com
To: asterisk-users@lists.digium.com
Date: Wed, 19 Aug 2009 11:07:21 -0400
Subject: Re: [asterisk-users] outbound calls not ringing
On Wed, 2009-08-19 at 13:54 +, Ott Rose wrote:
I
This has got to be an FAQ, so if someone can point me to where it is
answered, I would be greatly appreciated. The documentation for all this
stuff is scattered and (to me at least) either very sketchy or hard to
find.
What I want is a guide for how to convert from 1.4 with zaptel to 1.6
with
harry R escribió:
mysql -uasterisk -pasterisk asteriskdb
When I do that in a linux terminal it works.
But I always have this err 2002.
Greeting missing.
Elaborate missing.
Err 0x1b5a9f4c
You're not talking to machines here. :-)
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone
sip.conf
On Wed, 2009-08-19 at 15:55 +, Ott Rose wrote:
we are using Aastra 57i
i don't see that setting. where is it at?
From: jsulli...@opensourcedevel.com
To: asterisk-users@lists.digium.com
Date: Wed, 19 Aug 2009 11:07:21 -0400
Subject: Re: [asterisk-users] outbound calls
Can I make a non related suggestions?
Ditch Fedora and use CentOS.
Greg Woods wrote:
This has got to be an FAQ, so if someone can point me to where it is
answered, I would be greatly appreciated. The documentation for all this
stuff is scattered and (to me at least) either very sketchy or
On Monday 17 August 2009 18:03:10 pm Tilghman Lesher wrote:
How does one go about accessing gosub arguments from Asterisk in
extensions.lua?
You cannot. The various methods of dialplan creation are not
designed to be
interoperable. Some people have made various methods work (such
Dave Fullerton wrote:
I guess I just found a bug then, because the card above is a TE220B.
Here's a portion of the dmesg output:
You are correct sir; I wrote the code that was supposed to report the
VPM presence via dahdi_scan, but clearly did not test it properly
because it didn't work :-(
Hi,
On Wed, Aug 19, 2009 at 09:56:38AM -0600, Greg Woods wrote:
This has got to be an FAQ, so if someone can point me to where it is
answered, I would be greatly appreciated. The documentation for all this
stuff is scattered and (to me at least) either very sketchy or hard to
find.
What I
here is my sip.conf. i don't see it.
;;
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications
to ;
; this file must be done via the web gui. There are alternative files to make
;
; custom
I'm a 1.2 Luddite, but...
On Wed, 19 Aug 2009, Greg Woods wrote:
What I want is a guide for how to convert from 1.4 with zaptel to 1.6
with dahdi.
If you didn't find it on voip-info.org, please start an article.
All the recent kernel vulnerabilities are forcing me to upgrade my home
Oops! - You're using FreePBX - someone who knows more about FreePBX will
have to help you as I don't. May I also suggest that you bottom post in
future responses rather than top post; that makes it a little easier to
follow. Good luck - John
On Wed, 2009-08-19 at 16:59 +, Ott Rose wrote:
On Wednesday 19 August 2009 11:29:58 Brian Camp wrote:
On Monday 17 August 2009 18:03:10 pm Tilghman Lesher wrote:
How does one go about accessing gosub arguments from Asterisk in
extensions.lua?
You cannot. The various methods of dialplan creation are not
designed to be
On Wednesday 19 August 2009 10:43:37 Tilghman Lesher wrote:
On Wednesday 19 August 2009 05:54:32 Raimund Sacherer wrote:
I need CAP_FOWNER=ep for the asterisk process, i set it with setcap on
the file /usr/sbin/asterisk, it's there when i look on it with getcap,
but after starting and
On Wed, Aug 19, 2009 at 11:56 AM, Greg Woodsg...@gregandeva.net wrote:
This has got to be an FAQ, so if someone can point me to where it is
answered, I would be greatly appreciated. The documentation for all this
stuff is scattered and (to me at least) either very sketchy or hard to
find.
On Wed, 2009-08-19 at 12:25 -0400, Singer X.J. Wang wrote:
Can I make a non related suggestions?
Ditch Fedora and use CentOS.
Might be a possibility except that this is a catch-all home server. It
is used for things other than asterisk, so there are other reasons why I
need a more up-to-date
On Wed, 19 Aug 2009, Greg Woods wrote:
On Wed, 2009-08-19 at 12:25 -0400, Singer X.J. Wang wrote:
Can I make a non related suggestions?
Ditch Fedora and use CentOS.
Might be a possibility except that this is a catch-all home server. It
is used for things other than asterisk, so there are
On Wed, 2009-08-19 at 09:56 -0600, Greg Woods wrote:
This has got to be an FAQ, so if someone can point me to where it is
answered, I would be greatly appreciated. The documentation for all this
stuff is scattered and (to me at least) either very sketchy or hard to
find.
Well at least I'm
On Wed, 19 Aug 2009, Greg Woods wrote:
If at all possible I want to use the standard packaged version as it
makes security updates much easier.
I used to be a use the source Luke kind of guy. Now I'm a yum-aholic.
But, when the pain of using packages exceeds the hassle of the source,
I'll
Greg Woods wrote:
I heard one person say that extensions.conf should still work in 1.6; is
that true? Mine seems to be completely ignored. Is there some other
config file that I need to edit to make this happen? If I could do that,
I could get the OS switch done and it would buy me some time
Hi,
I am trying to implement a macro-screen mentioned at
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
I put the following code in my extensions_additional.conf
screen-from: You have a call;
screen-accept: Press 1 to accept this call or any other key to reject.;
Hi All,
I'm new to Asterisk, but am a relatively accomplished Linux guy
(RH5.1) .. I'm using the Asterisk-GUI to try and create a menu for
incoming calls on an Analog Trunk. I have recorded some .WAV files for
the menu, but when I try to upload the files, I get an AG101 message. So
I
On Wed, 2009-08-19 at 11:49 -0700, Steve Edwards wrote:
But, when the pain of using packages exceeds the hassle of the source,
I'll use the source without hesitation.
Agreed. I use trunk for MythTV, for instance, because there are features
not in the latest packaged version that I really do
On Wed, 2009-08-19 at 14:27 -0700, Steve Edwards wrote:
When the developers want to convert the config language, sooner or later
they will stop supporting the old stuff and it won't be possible to get
the newest supported features without converting.
I don't see that happening in my
On Wed, 19 Aug 2009, Greg Woods wrote:
When the developers want to convert the config language, sooner or later
they will stop supporting the old stuff and it won't be possible to get
the newest supported features without converting. So I don't want to
fall TOO far behind.
I don't see
Did you #include extensions_additional.conf in your extensions.conf file?
Verify this by doing dialplan show macro-screen from CLI.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bharath B.
Reddy Bynagari
Sent: Wednesday,
On Wed, Aug 19, 2009 at 03:48:22PM -0600, Greg Woods wrote:
This appears to be somewhat inconsistent in your determination to
update to 1.6. Are packages for 1.4 available for F11?
Well, Axel did just say that he might be providing some through ATrpms,
so that is a possibility.
Actually
At 02:27 PM 8/19/2009, you wrote:
Not at all. I'm just saying if the available packages are doing it for
you, compiling the source is pretty trivial. If the packages catch up
with F11 you can always install them then.
My first experience with Linux and Asterisk was putting a whatever
TrixBox was
Generally with FreePBX the ring options are set in the General Options -
you can set the Dial options which are normally tr, but I guess that
isn't working for you.
The SIP files you could edit would have custom in their name, otherwise
your changes will be overwritten when you reload freepbx
We have setup asterisk to handle our calls before between telco and an
Avaya definity. The PRI keeps locking up every so often.
In addition I keep getting this error when trying to call the avaya:
-- Channel 0/2, span 1 got hangup request, cause 102
-- Hungup 'Zap/2-1'
When that error
Yep, agreed.
Convert the file to the native codec(s) in which it will be played.
Alex, could you please elaborate on this? I am no audio guy.
On Media player, I can rip it into mp3 or wav or windows media audio.
Which one should I use?
___
--
Probably none of the ones you list, though I believe wav files are
uncompressed. Use SOX http://sox.sourceforge.net/ under Linux, Windows or
OSX and RIP/Convert the files to match the codec you are using for calls.
If you are accepting calls that use the GSM codec then have a set of MOH
files
On Thu, 20 Aug 2009, Lee, John (Sydney) wrote:
Convert the file to the native codec(s) in which it will be played.
Alex, could you please elaborate on this? I am no audio guy.
On Media player, I can rip it into mp3 or wav or windows media audio.
Which one should I use?
Neither.
If your
Hi All,
I'm new to Asterisk, but am a relatively accomplished Linux guy
(RH5.1) .. I'm using the Asterisk-GUI to try and create a menu for
incoming calls on an Analog Trunk. I have recorded some .WAV files for
the menu, but when I try to upload the files, I get an AG101 message. So
I
I have tried a lot like as
exten = 123,1,Dial(SIP/114SIP/113SIP/115)
and all the channels are dialing and if i answered any 3 of one, all the
channels except which one i answered are hung up..
I need all 3 channels are ringing and playback a message to any one or more.
So how to do it???
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