On Sat, 14 Apr 2007, Joe Acquisto wrote:
Tzafrir Cohen [EMAIL PROTECTED] Wrote: 4/14/2007 9:59 AM:
On Sat, Apr 14, 2007 at 08:58:57AM -0400, Joe Acquisto wrote:
Tzafrir Cohen [EMAIL PROTECTED] Wrote: 4/14/2007 7:41 AM:
Next-best thing would be to sart looking for differences in messages
On Sat, 2007-04-14 at 14:17 -0500, Lacy Moore - Aspendora wrote:
This was mentioned earlier:
I suspect IRQ Sharing.
I know. And I posted my /proc/interrupts showing that there were no
shared IRQ's.
And from the rest, it sounds like your network card and Digium card
are both sharing
It looks like swapping the soundcard and the Digium card would be a
good thing. From what I understand, the Digium cards should always
have the highest priority, and the IRQ with the highest number gets
the highest priority. Or do I have this backwards?
On 4/14/07, Greg Woods [EMAIL PROTECTED]
Hi all-
Equipment:
Xlite softphone
Polycom 301 with SIP 2.1.1 and BootROM 3.2.3
Polycom 501 with SIP 2.1.1 and BootROM 3.2.3
Asterisk 1.4.2
SIP Trunk to FWD
I wanted to post this problem as I haven't found it described in any of
the past presence threads on here.
I use an
On Sat, 2007-04-14 at 06:23 -0400, Joe Acquisto wrote:
A very simple question - what version is running?
A CLI - show version does not tell me, only shows info about who
compiled, and when. A google and other nefarious devices have not
yielded the secret.
Here is what worked for me:
Don't have an extra machine at the site. I'm starting to suspect the
FXO module or the telco. One FXO module hooked to another analog trunk
works flawlessly. When the client is in the office next, I'm going to
have them switch the cables.
Any other ideas in the meantime?
Thanks,
DR
Hi all,
I was trying to set up a conference room using the MeetMe
application and my asterisk is telling me that there is no MeetMe
application available for the extension I've dialed.
[Apr 14 20:57:11] WARNING[958] pbx.c: No application 'MeetMe' for
extension (internal, 600, 1)
So, I
Try this:
# strings -a /usr/sbin/asterisk | grep Digium
I get:
Asterisk 1.2.16, Copyright (C) 1999 - 2005, Digium, Inc. and others.
Asterisk 1.2.16, Copyright (C) 1999 - 2006 Digium, Inc. and others.
but if the version string has been removed from the sources, it's
Ronaldo Zacarias Afonso wrote:
Hi all,
I was trying to set up a conference room using the MeetMe
application and my asterisk is telling me that there is no MeetMe
application available for the extension I've dialed.
[Apr 14 20:57:11] WARNING[958] pbx.c: No application 'MeetMe' for
I have a trunk setup to broadvoice and it seems to work fine. I needed
an indial in another localtion so setup a second trunk on a new bv
account.
Either account regiteres by itself but when both are enabled, it sits in
Request Sent status on sip show registry
anyone else had this issue ? I
HI all,
I have a queue say 5000 and there are 10 member in the queue. When
there is a call to the queue, the members will ring according to the
defined strategy. In day end, I have to create a report about the
queue and its member. But I found that it is very difficult to find
the relation
hello,
I use both * 1.4 and *NOW... because the *gui is incomplete in *NOW, I
loaded 1.4 over *NOW because the gui regenerates files that, well, don't
seem to work very well. it seems to me the gui creates the users.conf
file, and then a script creates or uses the users.conf to create the
Hi All,
Customer is requesting 1 incoming toll free #, that dial out to 4
different terminating numbers, not ring all at once but ring #1, then
#2, then #3, then #4, then back to #1 consecutively on inbound calls,
regardless if someone is on #1. So this is not like a hunt group,
more like
You will probably find what you are looking for here:
/var/log/asterisk/queue_log
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rilawich
Ango
Sent: Saturday, April 14, 2007 21:07
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
yusuf wrote:
Hi,
as far as I know, it only says ignoring when you do a reload, as
Asterisk is telling you its not reconfiguring this variable, to change
it you might need a restart. So hanguponpolarityswitch only gets looked
at on startup, not reloads.
Thanks for that. I restarted and I
Pezhman Lali wrote:
Dear
I am using Ser+Asterisk, for sip providing.
there is a problem,
the asterisk does not return back the busy tones to
the sip phones.
for example, if the destination number is busy, we
are hearing waiting ring from sip phones, and after
60sec(timeout) the call will
It can get confusing at first but once you get the hang of it, its a breeze.
first take a look at: http://www.voip-info.org/wiki-Asterisk+cmd+MYSQL
there is plenty of (correct)info and (working)examples on this page.
you have to \escape a space,quote and double quote, comma and backslash - so
On Fri, 13 Apr 2007, Ryan Stille wrote:
I've read through the Wiki, I know its hard to nail down hardware
requirements because it really depends on what you are going to do. I'm very
new to Asterisk, haven't even read my Asterisk for Dummies book yet (its in
the mail).
Can you tell me if
Hello,
Can you tell more about Xen? I would like to install Debian Etch with
Xen and use A Digium 4-port E1 in a guest domain. Is it possible? I
read of much problems with cards in a guest domain.
I have Xen running with DNS-server/ Web-server guests, also a VoIP
only Asterisk, but a telephony
A very simple question - what version is running?
A CLI - show version does not tell me, only shows info about who compiled, and
when. A google and other nefarious devices have not yielded the secret.
Sure, I could scour the docs and determine what is features do not exist in all
versions,
On Sat, 14 Apr 2007, Joe Acquisto wrote:
A very simple question - what version is running?
A CLI - show version does not tell me, only shows info about who
compiled, and when. A google and other nefarious devices have not
yielded the secret.
Are you sure?
I get:
dsx*CLI show version
Gordon Henderson [EMAIL PROTECTED] Wrote: 4/14/2007 7:02 AM:
On Sat, 14 Apr 2007, Joe Acquisto wrote:
A very simple question - what version is running?
A CLI - show version does not tell me, only shows info about who
compiled, and when. A google and other nefarious devices have not
On Sat, Apr 14, 2007 at 07:15:36AM -0400, Joe Acquisto wrote:
Gordon Henderson [EMAIL PROTECTED] Wrote: 4/14/2007 7:02 AM:
On Sat, 14 Apr 2007, Joe Acquisto wrote:
A very simple question - what version is running?
A CLI - show version does not tell me, only shows info about who
On Fri, Apr 13, 2007 at 03:47:15PM -0400, Jerry Geis wrote:
I am getting the following compile error on centos 5. Any suggestions?
Jerry
I can't tell you waht went wrong in your build, but there are apckages
of zaptel available at http://atrpms.net/dist/el5/zaptel/
CC [M]
Hi,
Actually this was the problem:
debian etch, asterisk 1.4.2, zaptel 1.4.1, ... cannot find the
chan_zap.so module file anywhere, tried recompiling with zaptel
1.4.0... no change...
Here is a solution from:
http://www.asteriskguru.com/archives/image-vp266527.html
Compile asterisk 1.4.1
Tzafrir Cohen [EMAIL PROTECTED] Wrote: 4/14/2007 7:41 AM:
On Sat, Apr 14, 2007 at 07:15:36AM -0400, Joe Acquisto wrote:
Gordon Henderson [EMAIL PROTECTED] Wrote: 4/14/2007 7:02 AM:
On Sat, 14 Apr 2007, Joe Acquisto wrote:
A very simple question - what version is running?
A CLI - show
On Sat, Apr 14, 2007 at 08:58:57AM -0400, Joe Acquisto wrote:
Tzafrir Cohen [EMAIL PROTECTED] Wrote: 4/14/2007 7:41 AM:
Next-best thing would be to sart looking for differences in messages and
such from recent commits.
I've only this one version to check, at this time.
1.2 or
What you are describing is only available in a guest domain if your CPU(s)
support hardware virtualization. If they do, however, this configuration is
pretty straight forward.
Xen as a virtualizing solution ships in a well-documented format in the Fedora
6 distribution. If you would prefer
Hello, I'm using two Cisco 7960 phones currently loaded and showing
Firmware POS3-07-4-0 (Version 7.4?) and I'm having a strange problem.
Whenever the phone is supposed to try to load anything over HTTP from
my Apache 2.2.x web server, the connection just sits and times out.
Nothing shows up in
On Sat, Apr 14, 2007 at 01:58:22PM +0200, Axel Thimm wrote:
On Fri, Apr 13, 2007 at 03:47:15PM -0400, Jerry Geis wrote:
I am getting the following compile error on centos 5. Any suggestions?
Jerry
I can't tell you waht went wrong in your build, but there are apckages
of zaptel available
I'm getting a random fast busy signal after two rings on an incoming ZAP
channel through an FXO interface on a TDM400P. I have a NoOp in the
dial plan showing the callerID. When the random fastbusy happens, no
CallerID is sent. When it works, the CallerID information is sent. Is
this a card
Can you put a protocol analyzer on your E1 and send us the layer 3 trace?
John Treble
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hi
My a2billing adds |HrL automatically to dial string,
I can not find the source of this task,
I need to remove r from all dial strings,
Thanks for your help.
Best
Mani
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection
Did you try taking NoOP out?does it work fine then? What version of
Linux are you using and what version of asterisk?
Astawerks
VoIP Hardware sales and Consulting
HYPERLINK http://www.astawerks.com/www.astawerks.com
614-495-1400
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL
I did take the NoOp out and there was no difference.
*1.2.17 and zaptel 1.2.16.
Thanks!
DR
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug
Segalowitz
Sent: Saturday, April 14, 2007 1:30 PM
To: 'Asterisk Users Mailing List -
On Fri, 2007-04-13 at 11:36 -0800, Mojo with Horan Company, LLC wrote:
in some
places, there are so few phone prefixes, you can simply match them
exactly.
Thanks for that idea; that should be even easier to do here, because we
live in 10-digit land and have two overlapping area codes,
Pezhman Lali wrote:
hi
My a2billing adds |HrL automatically to dial string,
I can not find the source of this task,
I need to remove r from all dial strings,
Thanks for your help.
Best
Mani
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Tired of spam? Yahoo! Mail has the
You can correct de dialing options in a2billing.conf file. Almost at the end
of the file.
etc/asterisk/a2billing.conf
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pezhman Lali
Sent: Saturday, April 14, 2007 12:10 PM
To: [EMAIL PROTECTED]
Subject:
for kicks lets try setting it up on another machine, if you have an extra
one and see if it still does it.
Astawerks
VoIP Hardware sales and consulting
HYPERLINK http://www.astawerks.com/http://www.astawerks.com
614-495-1400
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
On 4/13/07, Mike [EMAIL PROTECTED] wrote:
True, but that being said 1.6.7 did re-register every 30 seconds with no
issues.
Did the phone loss in performance after the upgrade?
Not for me. Did not observe any lesser performance.
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On 4/14/07, Greg Woods [EMAIL PROTECTED] wrote:
Even worse, I discovered
that the same problem affects racoon/ipsec-tools as well; I get racoon
errors in the log about hash mismatches and messages too short. Unload
the zaptel drivers, and the tunnel is established immediately. I was
hoping to
On 4/14/07, Greg Woods [EMAIL PROTECTED] wrote:
On a possibly related note, I find that I cannot build the Zaptel
drivers at all on newer FC6 kernels. I am running 2.6.19-1.2911.6.5.fc6
Never mind about my previous message about compiling Zaptel. It's
unrelated, but what may be related is
Asterisk works great with openvz. Ive run 4 VE's with combined average
around 32 simultaneous calls at any time and you wouldn't know the
difference.
On 4/13/07, Voip Asterisk [EMAIL PROTECTED] wrote:
Anyone here running asterisk on openvz, if so what are your experiences?
Right now we are
Tzafrir Cohen [EMAIL PROTECTED] Wrote: 4/14/2007 9:59 AM:
On Sat, Apr 14, 2007 at 08:58:57AM -0400, Joe Acquisto wrote:
Tzafrir Cohen [EMAIL PROTECTED] Wrote: 4/14/2007 7:41 AM:
Next-best thing would be to sart looking for differences in messages and
such from recent commits.
On Sat, Apr 14, 2007 at 04:22:25PM -0400, Joe Acquisto wrote:
Tzafrir Cohen [EMAIL PROTECTED] Wrote: 4/14/2007 9:59 AM:
On Sat, Apr 14, 2007 at 08:58:57AM -0400, Joe Acquisto wrote:
Tzafrir Cohen [EMAIL PROTECTED] Wrote: 4/14/2007 7:41 AM:
Next-best thing would be to sart
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