On Sun, 22 Feb 2009, Doug wrote:
This would work:
~~
Codegen case:
http://www.newegg.com/Product/Product.aspx?Item=N82E16811182566
Thermaltake power supply:
http://www.mwave.com/mwave/viewspec_v2.asp?scriteria=BA23480
Motherboard:
On Mon, Feb 23, 2009 at 08:00:38AM +, Gordon Henderson wrote:
On Sun, 22 Feb 2009, Doug wrote:
Interesting shopping list - I've just built a new server for my co-lo and
it's an Intel Atom mobo. Normally I do use AMD though, but right now,
power consumption is an issue and hen AMD's low
Dear Sir,
Kindly note that the problem is on command $AGI-get_variable('
variablename');
The AGI seems that it's not reading nothing from asterisk
Regards
On Mon, Feb 23, 2009 at 9:26 AM, Yawar Hadi yawarh...@gmail.com wrote:
ohh got it...sorry for miss interpretation
On Mon, Feb
Hi:
I want to receive a fax with an E1 link connected to A102d card from a
fax machine,but after dialling the phone number, it connects then will
be busy.In fact asterisk can't detect the fax.These are zapata.conf,
extensions.conf filels and debug in console:
extensions.conf:
[from-pstn]
On Mon, 23 Feb 2009, Tzafrir Cohen wrote:
On Mon, Feb 23, 2009 at 08:00:38AM +, Gordon Henderson wrote:
On Sun, 22 Feb 2009, Doug wrote:
Interesting shopping list - I've just built a new server for my co-lo and
it's an Intel Atom mobo. Normally I do use AMD though, but right now,
power
i got the problem.
i am looking for a wrong variable.
if u want to read variables from astersk with $AGI-get_variable() command
then write the variable name as mentioned on voip-info site.
so if u want to read extension then supplu variable name like
$myno=$AGI-get_variable('EXTEN');
Tilghman Lesher schrieb:
On Wednesday 11 February 2009 13:39:58 Philipp Kempgen wrote:
Tilghman Lesher schrieb:
On Wed, 11 Feb 2009, Tilghman Lesher wrote:
My viewpoint is that you should work on separation of your application
code versus data, so that other than new development, your
Hello Hadi,
EXTEN is working fine and I can get the dialed number...Now suppose I need
to fetche the extension username from where I'm dialing...What is the
variable name here?Or where i can find Variables name...Can you give me a
link where these variables are defined?
Regards
On Mon, Feb 23,
Hi,
I know this is not a 100% Asterisk question, but is there anyone who has the
Flash Operator Panel working with Asterisk 1.6??
In asternic.org there is a version that show call status but you cant make
transfers or originate a call.
Has anyone fixed the op_server.pl file to fully work with
Pablo Bernasconi wrote:
Has anyone fixed the op_server.pl file to fully work with Asterisk 1.6???
Nicolas is looking for beta testers for the version that works with 1.6.
I'd suggest that you subscribe to FOP mailing.
Doug
--
Ben Franklin quote:
Those who would give up Essential
Have you tried this?
'su - asterisk
'cd /var/lib/asterisk/moh
When this works, so will *.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph L.
Casale
Sent: Sunday, February 22, 2009 11:06 PM
To:
Have you tried this?
'su - asterisk
'cd /var/lib/asterisk/moh
When this works, so will *.
Yup, I should have stated that more specifically:
# ll /var/lib/asterisk/moh
total 6604
-rw-r- 1 asterisk asterisk 1939794 Sep 20 2006 fpm-calm-river.wav
-rw-r- 1 asterisk asterisk 2582196 Sep 20
Asterisk/Skype update available here -
http://blogs.digium.com/2009/02/23/skype-for-asterisk-update/
It's definitely an update that updates absolutely nothing :-), more
news at 11 :P
Cheers,
Dean
___
-- Bandwidth
We have 40 nortel ip2002 phones connected to asterisk 1.6, the problem I have
is that the phone looses the connections with the server and then drops calls,
we can reconnect but the customers don't like it.
Anyone has the same problem?
/ralf
In my directory these files are 0644 (-rw-r--r--) and directory perms are
0755 (-rwxr-x-wx). Check that.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph L.
Casale
Sent: Monday, February 23, 2009 8:55 AM
On Mon, Feb 23, 2009 at 12:06 AM, Joseph L. Casale
jcas...@activenetwerx.com wrote:
I am running Asterisk as non root and have set the required permissions for
all
directories including the moh dir specified in musiconhold.conf yet asterisk
still complains it doesn't have access when
Danny Nicholas schrieb:
In my directory these files are 0644 (-rw-r--r--) and directory perms are
0755 (-rwxr-x-wx)
0755 is rwxr-xr-x :-)
Philipp Kempgen
--
AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de
Asterisk: http://the-asterisk-book.com -
Hi all,
We are having troubles with the 302 responses handling by asterisk 1.4. The
302 SIP responses generates an INVITE in which To and R-URI are the same,
when in the RFC 3261 (8.1.3.4) RECOMMENDS reusing of the same To, From and
Call-ID values than the original request. Is there any
On Mon, Feb 16, 2009 at 2:29 PM, Asterisk Asterisk
nt_aster...@yahoo.com wrote:
I need your help: please help test the gender detection module at
575-613-4392.
I wrote a gender detection module and thought I'd try it out. It only takes
a second. I've been showing 90%+ accuracy and I want
to
On Mon, 23 Feb 2009, michel freiha wrote:
EXTEN is working fine and I can get the dialed number...Now suppose I
need to fetche the extension username from where I'm dialing...What is
the variable name here?Or where i can find Variables name...Can you give
me a link where these variables
On Mon, Feb 23, 2009 at 3:30 AM, fateme fatah faza_...@yahoo.com wrote:
extensions.conf:
[from-pstn]
exten = 9711315,1,Answer()
exten = 9711315,2,Wait(10)
Why on earth are you waiting TEN seconds to actually receive the fax?
Have you tried ripping that out of your dialplan?
exten =
On Mon, 23 Feb 2009, Yawar Hadi wrote:
so if u want to read extension then supplu variable name like
$myno=$AGI-get_variable('EXTEN');
hope u get it
I think passing the variable on the command line is more flexible than a
fixed variable, particularly the much more limiting ${EXTEN}.
My
We are proud to release Astlinux 0.6.3. All users of AstLinux should
upgrade to this release. Files are available for download at the
Astlinux SourceForge project page.
https://sourceforge.net/project/showfiles.php?group_id=170462
Updates include new versions of Asterisk, Asterisk-gui,
Leonardo Gomes Figueira wrote:
Do I have to contact Digium to reset the license key even if I'm not
trying to register a new MAC ? I just want an updated license key file.
I'd like to help you debug this, so we can improve our registration system.
Can you send me the existing license file
On Monday 23 February 2009 04:49:15 Philipp Kempgen wrote:
Tilghman Lesher schrieb:
On Wednesday 11 February 2009 13:39:58 Philipp Kempgen wrote:
Tilghman Lesher schrieb:
On Wed, 11 Feb 2009, Tilghman Lesher wrote:
My viewpoint is that you should work on separation of your
How to Delete all files under folder in CENTOS ?
Need urgent help thats why mailing here
Excuse me for OTP.
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rm * works ?
On Mon, Feb 23, 2009 at 11:07 PM, David @ULC ucoms2...@gmail.com wrote:
How to Delete all files under folder in CENTOS ?
Need urgent help thats why mailing here
Excuse me for OTP.
___
-- Bandwidth and Colocation Provided by
On Mon, Feb 23, 2009 at 6:37 PM, David @ULC ucoms2...@gmail.com wrote:
How to Delete all files under folder in CENTOS ?
http://lmgtfy.com/?q=how+to+recursively+remove+files+in+a+folder+in+linux
___
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On Mon, Feb 23, 2009 at 11:07:57PM +0530, David @ULC wrote:
How to Delete all files under folder in CENTOS ?
Need urgent help thats why mailing here
Excuse me for OTP.
rm -rf CENTOS/folder/*
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
On Mon, 23 Feb 2009, David @ULC wrote:
How to Delete all files under folder in CENTOS ?
Need urgent help thats why mailing here
Excuse me for OTP.
Please find a more appropriate forum for beginning Linux|Unix user
questions.
Thanks in advance,
# cd /directory
# rm -rf *
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
: http://www.asterisk-jobs.com
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David @ULC
Sent: Monday, February 23,
Brian Degenhardt wrote:
Can somebody comment on how the RFC differs from any current or planned
features in the IAX2 protocol? For example, changes to IAX2 encryption,
media-only transfers, or any plans on changing codec negotiation to
support more options.
Asterisk 1.6 releases support
On Mon, Feb 23, 2009 at 06:48:42PM +0100, Michael Iedema wrote:
On Mon, Feb 23, 2009 at 6:37 PM, David @ULC ucoms2...@gmail.com wrote:
How to Delete all files under folder in CENTOS ?
http://lmgtfy.com/?q=how+to+recursively+remove+files+in+a+folder+in+linux
You have to know the right
FreePBX in a Cloud With a Click
(http://voxilla.com/2009/02/23/freepbx-in-a-cloud-with-a-click-1436
)
Since making freely available Voxilla’s “pre-built” Asterisk
installation for Amazon’s Elastic Compute Cloud (EC2) (Asterisk on the
Cloud With a Click), we’ve received many requests for
The specific error is that it cannot chdir to the music on hold
directory. Are you sure you have the right directory?
what do you get when you do:
CLI moh show files
and
CLI moh show classes
The specific error is that it cannot chdir to the music on hold
directory. Who owns the parent directory?
format c:
:-P
2009/2/23 Tzafrir Cohen tzafrir.co...@xorcom.com
On Mon, Feb 23, 2009 at 06:48:42PM +0100, Michael Iedema wrote:
On Mon, Feb 23, 2009 at 6:37 PM, David @ULC ucoms2...@gmail.com wrote:
How to Delete all files under folder in CENTOS ?
Anyone using 5ess with success? I have a customer that wants to use it.
Seems to WORK for a short time then there are errors on the D-channel
and their side (nortel switch) shuts down the channel.
They were saying something about a slip error. I dont know this term.
I am using dahdi 2.1.0.4 and
What are they using for the clock source?
--Don
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Monday, February 23, 2009 1:13 PM
To: asterisk-users@lists.digium.com
Subject:
NVLineDetect , I dont find it in the web for asterisk 1.4
Anybody has a link that works?
*---*
*-Edwin Quijada
*-Developer DataBase
*-JQ Microsistemas
*-809-849-8087
* Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo
Anyone using this feature of asterisk's voicemail? I'd never heard of
ADSI, and saw it as I was perusing the voicemail source this morning. Is
it some kind of visual way of managing voicemail on your phone's display,
or does it require a terminal of some kind?
Thanks!
j
On Mon, Feb 23, 2009 at 05:11:05PM -0200, David fire wrote:
format c:
'mformat c:' would work if you have mtools installed.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com
Jeff LaCoursiere wrote:
Anyone using this feature of asterisk's voicemail? I'd never heard of
ADSI, and saw it as I was perusing the voicemail source this morning. Is
it some kind of visual way of managing voicemail on your phone's display,
or does it require a terminal of some kind?
It
ADSI is Analog Display Services Interface.
Haven't used it with Asterisk, but ADSI-enabled phones interact with an
analog switch using FSK.
--Don
Don Kelly
PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax
-Original Message-
From:
The Asterisk.org development team is proud to announce the release of
Asterisk 1.6.0.6. This release is available for download from
http://downloads.digium.com/.
This release is a significant bug fix update for the 1.6.0 release series.
In addition, this release is recommended for all users of
Is there some magic to compiling asterisk-addons-1.6.0 under Debian 2.6.18
and mysql 5.0? I am unable to get configure to recognize the existance of
mysqlclient. Imparticular, when it gets to:
checking for mysql_init in -lmysqlclient... it returns no.
For the past several releases, I've had to
On Mon, Feb 23, 2009 at 12:26:09PM -0800, Bruce Komito wrote:
Is there some magic to compiling asterisk-addons-1.6.0 under Debian 2.6.18
and mysql 5.0? I am unable to get configure to recognize the existance of
mysqlclient. Imparticular, when it gets to:
checking for mysql_init in
Hi people!
I am not getting really smart. I get the SVN Edition of asterisk GUI
interface, compiled and love to get it to run, what won't work. What am
I doing wrong?!
svn checkout http://svn.digium.com/svn/asterisk-gui/branches/2.0
make
make checkconfig
make install
and If I open one of the
That was the silver bullet...thanks!
Bruce Komito
WPTI Telecom
(775) 236-5815
On Mon, 23 Feb 2009, Tzafrir Cohen wrote:
On Mon, Feb 23, 2009 at 12:26:09PM -0800, Bruce Komito wrote:
Is there some magic to compiling asterisk-addons-1.6.0 under Debian 2.6.18
and mysql 5.0? I am unable to
Asterisk is not a localhost function. It binds to the IP address of the
machine (unless you replace 0.0.0.0 with 127.0.0.1).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tamer Higazi
Sent: Monday, February
Does anyone know why?
ThePBX*CLI
-- Executing [310-456-7...@from-trunk:1]
Set(SIP/202.101.202.101-b763ce60, __FROM_DID=310-456-7890) in new stack
-- Executing [310-456-7...@from-trunk:2]
ExecIf(SIP/202.101.202.101-b763ce60, 1
|Set|CALLERID(name)=310-456-0987) in new stack
-- Executing
Hi people!
I am not getting really smart. I get the SVN Edition of asterisk GUI
interface, compiled and love to get it to run, what won't work. What am
I doing wrong?!
svn checkout http://svn.digium.com/svn/asterisk-gui/branches/2.0
make
make checkconfig
make install
and If I open one of the
What are they using for the clock source?
--Don
Don,
The digium card is setup for NET. they are cpe.
Jerry
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On 2/20/09, John Todd jt...@digium.com wrote:
Mark and Ed received word today that the long-awaited RFC for IAX2 has
been approved by the IETF, and is now published:
http://www.rfc-editor.org/authors/rfc5456.txt
Thanks to Ed Guy, Mark Spencer, Brian Capouch, Frank Miller, and Kenny
On Mon, Feb 23, 2009 at 03:12:26PM -0600, Danny Nicholas wrote:
Asterisk is not a localhost function. It binds to the IP address of the
machine (unless you replace 0.0.0.0 with 127.0.0.1).
Huh? By default it binds to 0.0.0.0 (all interfaces)
But then again, this is easy to test:
netstat
This is perhaps so. But I still think it is better form to use the local IP
as opposed to localhost or 127.0.0.1 especially if you might have more than
1 *.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
I found that after moving to Asterisk 1.6 and the latest SVN of
ASterisk-GUI, the link changed from:
http://localhost:8088/asterisk/static/config/cfgbasic.html
to:
http://localhost:8088/static/config/cfgbasic.html
I don't know if you'll find the same or not...
Bob
On Mon, 2009-02-23 at 22:17
At least on 1.4, Asterisk in the path is a conf option (http or *, don't
remember which).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bob Pierce
Sent: Monday, February 23, 2009 4:25 PM
To: Asterisk Users
On Mon, Feb 23, 2009 at 2:13 PM, Jerry Geis ge...@pagestation.com wrote:
Anyone using 5ess with success? I have a customer that wants to use it.
Seems to WORK for a short time then there are errors on the D-channel
and their side (nortel switch) shuts down the channel.
They were saying
Has anyone written a Windows Mobile app which gets the MWI info from a SIP
server, and updates the VM counter in the OS?
I'd like my PPC to show my voicemail count (and SIP MWI seems like the
easiest way)
___
-- Bandwidth and Colocation
On Mon, Feb 23, 2009 at 04:25:13PM -0600, Danny Nicholas wrote:
This is perhaps so. But I still think it is better form to use the local IP
as opposed to localhost or 127.0.0.1 especially if you might have more than
1 *.
Why?
One obvious problem with it: one cannot easily copy paste links
I should get all my non-human friends to call this number...or do you
want to call non-humans?
PaulH
Edwin Quijada wrote:
NVLineDetect , I dont find it in the web for asterisk 1.4
Anybody has a link that works?
*---*
*-Edwin Quijada
Hello guys,
I recently observed that my asterisk sends me sms like messages on my
phone (Nokia E71), I mean is SMS but is delivered some kind in-band
though VoIP. Is strange because this messages contains informations
about my voicemail and is sent by voicem...@mydomainxxx.com. I noticed
that this
On Mon, 2009-02-23 at 17:13 -0500, Jerry Geis wrote:
What are they using for the clock source?
The digium card is setup for NET. they are cpe.
I think you're confused here... NET and CPE don't control the clock
source, they simply specify which end of the connection is the network
you are getting the info about the voicemail becausethe soft on your phone
support it.
in sip.conf you can find some parameters to send that info.
in other soft phones like x-lite you will have the same info.
David
2009/2/23 Catalin S. jonsonpla...@gmail.com
Hello guys,
I recently observed
Dear all, I have Asterisk 1.4 with SIP. I have a voicemail implemented
with GSM sound files.
The problem is I have IP phones Utopix HyperPhone 202 which support
only G.729a/u and G.723.1 high/low, but not GSM.
If I choose G.729A the pass-throu calls among users are OK, but
Asterisk can't
Dear All,
I would really need to thank you all for the great help that I got here from
all of you and specially Mr. Yawar hadi for his great assist and
professionalism
Thanks
On Mon, Feb 23, 2009 at 6:48 PM, Steve Edwards asterisk@sedwards.comwrote:
On Mon, 23 Feb 2009, Yawar Hadi wrote:
Hi!
I have problems building asterisk 1.6.0.6.
./configure --prefix=/usr
make
gets me:
enerating embedded module rules ...
[CC] extconf.c - extconf.o
In file included from /usr/local/include/datatypes.h:50,
from /usr/local/include/err.h:49,
from
On Mon, 23 Feb 2009, Alejandro Cabrera Obed wrote:
Dear all, I have Asterisk 1.4 with SIP. I have a voicemail implemented
with GSM sound files.
The problem is I have IP phones Utopix HyperPhone 202 which support only
G.729a/u and G.723.1 high/low, but not GSM.
g723 is supported by
is any chance to use this feature to send messages on this kind of phones?
On Tue, Feb 24, 2009 at 1:39 AM, David fire ddf...@gmail.com wrote:
you are getting the info about the voicemail becausethe soft on your phone
support it.
in sip.conf you can find some parameters to send that info.
in
Are you sure this is not just a standard SIP MWI message?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Catalin S.
Sent: February 23, 2009 8:01 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] strange
I am running Asterisk 1.4.22.2, though I have also found this problem with
1.4.23.x
Sometimes after I hang up the system continues to spew packets to my phone
causing it to become unusable until I restart Asterisk.
Michael
___
-- Bandwidth and
I have 320 GB SATA HDD.
When I checked my phpsysinfo, it shows 95% HDD is filled.
[r...@vicidialnow ~]# df
Filesystem 1K-blocks Used Available Use% Mounted on
/dev/sda2 301924504 285002780 1337472 100% /
/dev/sda1 101086 11062 84805 12% /boot
tmpfs 1553832 0 1553832 0% /dev/shm
[r...@vicidialnow
On Mon, Feb 23, 2009 at 03:10, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Mon, Feb 23, 2009 at 08:00:38AM +, Gordon Henderson wrote:
On Sun, 22 Feb 2009, Doug wrote:
Interesting shopping list - I've just built a new server for my co-lo and
it's an Intel Atom mobo. Normally I do use
Just restarting it won't do anything. You could use the following
command to find any files over 200mb on the system. Be careful about
blindly deleting stuff though
*find / -type f -size +200M
Darren Wiebe
dar...@aleph-com.net
*
David @ULC wrote:
I have 320 GB SATA HDD.
When I
try format c:
and get a book about linux! or a course!
this is an asterisk mailing list not a linux/centos/vicidial now mailing
list.
and to solve that you need to delete some files.
probably the recorded audios.
cd /some dir/maybe other dir
rm * -r -f
PLEASE DONT DO THIS AT THE ROOT DIR OR
[r...@vicidialnow ~]# find / -type f -size +200M
/proc/kcore
/var/log/httpd/error_log.3
/var/log/httpd/access_log.3
/var/log/httpd/access_log.1
/var/log/httpd/error_log.2
/var/log/httpd/error_log.4
/var/log/httpd/access_log.2
/var/log/httpd/access_log.4
/var/log/httpd/error_log.1
I believe I can
On Tue, 24 Feb 2009, David @ULC wrote:
I have 320 GB SATA HDD.
When I checked my phpsysinfo, it shows 95% HDD is filled.
Please find a more appropriate forum for beginning Linux|Unix user
questions.
Thanks in advance,
I have a new Polycom Spectralink 8002 and am having trouble with the
configuration or the unit but I can't see what's wrong. The unit does
not seem to even attempt to register with the Asterisk proxy but I can
make calls to it. I have viewed the syslog from the device which it
will actually
I have been using the inbound 800 services from vitelity. Slowly the
usage has been rising and in the month of Jan the bill was for $650. I
am currently on a 1.9 cents a minute plan. Am I paying too much ?
Some suggestions my team generated to reduce the toll free incoming
call bill were:
1.
When I am trying to delete voice logs,
[r...@vicidialnow monitor]# rm * -r -f
-bash: /bin/rm: Argument list too long
[r...@vicidialnow monitor]#
Argument list too long is coming as a road block.
Now way to forcefully delete files ?
On Tue, Feb 24, 2009 at 7:30 AM, David @ULC ucoms2...@gmail.com
ko gui nua
--
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My server is down :-(
Thats why posted here
On Tue, Feb 24, 2009 at 7:05 AM, David @ULC ucoms2...@gmail.com wrote:
I have 320 GB SATA HDD.
When I checked my phpsysinfo, it shows 95% HDD is filled.
[r...@vicidialnow ~]# df
Filesystem 1K-blocks Used Available Use% Mounted on
/dev/sda2
On Tue, 24 Feb 2009, David @ULC wrote:
When I am trying to delete voice logs,
[r...@vicidialnow monitor]# rm * -r -f
-bash: /bin/rm: Argument list too long
In the past 30 days, you've asked questions about
configuring Apache to process PHP files,
Vicidial,
Ntework Cards,
Auto Detecting
On Mon, 23 Feb 2009, David fire wrote:
[snip]
cd /some dir/maybe other dir
rm * -r -f
PLEASE DONT DO THIS AT THE ROOT DIR OR YOU WILL ERASE ALL THE DISK.
CD TO THE TARGET DIRECTORY OR YOU WILL DESTROY YOUR SERVER.
Nah, you will only get as far as some shared libraries before the system
go to the previos dir and do
ls -lha target dir
write somewhere the owner and the attributes
rm target dir -r -f
you will erase all the target directory and all the sub directorys.
then
mkdir target dir
chmod atributes target dir
chown owner.group target dir
if you have problems use man
hi
in argentina we have 810 you pay only the long distant and the caller pay
the local rate.
if the caller is in your same city you dont pay anything.
David
2009/2/24 Vikas topg...@gmail.com
I have been using the inbound 800 services from vitelity. Slowly the
usage has been rising and in the
I find it a little strange that for some reason your box is using includes
located in /usr/local... while there could be reason for this, that seems
like a sign that something might be a little broken on your box.
Also, if you don;t mind me asking...
why would you want to install * directly in
On Mon, 23 Feb 2009, Vikas wrote:
I have been using the inbound 800 services from vitelity. Slowly the
usage has been rising and in the month of Jan the bill was for $650. I
am currently on a 1.9 cents a minute plan. Am I paying too much ?
Some suggestions my team generated to reduce the
On Mon, Feb 23, 2009 at 9:38 PM, Steve Edwards asterisk@sedwards.comwrote:
On Tue, 24 Feb 2009, David @ULC wrote:
When I am trying to delete voice logs,
[r...@vicidialnow monitor]# rm * -r -f
-bash: /bin/rm: Argument list too long
In the past 30 days, you've asked questions about
Wow that's crazy, 1.9 is pretty much as good as your going to get. I would
find out where were the most of your traffic is coming from and get local
numbers in those areas. When the person calls your 1800 number check if
there is a local number for them to use if so play the message with the
local
On Mon, Feb 23, 2009 at 9:10 PM, Vikas topg...@gmail.com wrote:
1. When people call in on the 800 number take the local number they
are calling from and then call them back from our unlimited outgoing
account from broadvoice.
I would recommend IVR-ing this as an option, on the premise that
On Mon, 23 Feb 2009, Vikas wrote:
I have been using the inbound 800 services from vitelity. Slowly the
usage has been rising and in the month of Jan the bill was for $650. I
am currently on a 1.9 cents a minute plan. Am I paying too much ?
Some suggestions my team generated to reduce the
The ATA186I1 is way past end of life,
it was the first version of the ATA186,
but it's still working with only a couple
adjustments to a factory default configuration.
Unlike the brand spanking new Cisco/Linksys SPA2102.
Which has a number outstanding issues with Cisco/Linksys
tech support.
On Mon, Feb 23, 2009 at 9:10 PM, Vikas topg...@gmail.com wrote:
I have been using the inbound 800 services from vitelity. Slowly the
usage has been rising and in the month of Jan the bill was for $650. I
am currently on a 1.9 cents a minute plan. Am I paying too much ?
I don't pay the bill, so
At 15:16 2/23/2009, Doug wrote:
Does anyone know why?
The problem was that I had moved the box
to a different IP address.
Needed to change the following setting in:
sip_general_custom.conf
externip=The.New.Actual.IP
___
-- Bandwidth and
If you are being paid to work on an Asterisk system, you are in over your
head. You are defrauding your boss and most likely will give him and
everyone in the company a bad impression of Asterisk.
Continuing to answer your questions will only continue to enable you.
Please take a step
Steve Edwards wrote:
On Tue, 24 Feb 2009, David @ULC wrote:
When I am trying to delete voice logs,
[r...@vicidialnow monitor]# rm * -r -f
-bash: /bin/rm: Argument list too long
In the past 30 days, you've asked questions about
configuring Apache to process PHP files,
Vicidial,
My asterisk box is in a datacenter in San Francisco CA.
The vitelity servers are 30 ms away from my asterisk server.
Lets consider the scenario that some one is calling from San Francisco
to the 800 number which I have with Vitelity.
My question is:
Does the audio first flow to the vitelity
,must visit this site for more info
http://www.voip-info.org/wiki-Asterisk+AGI
and also take help from google its a good source
On Tue, Feb 24, 2009 at 5:08 AM, michel freiha mich...@gmail.com wrote:
Dear All,
I would really need to thank you all for the great help that I got here
from all
Hello,
I'm looking for a free US DID in either North Carolina or Georgia State.
I've browsed the list on voip-users, but couldn't really find anything
(maybe I'm blind).
Does anyone know a provider?
Thanks,
Tobias
___
-- Bandwidth and Colocation
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