vitelity.
Thanks,
--Warren Selby
On Oct 9, 2009, at 4:48 PM, Ken D'Ambrosio k...@jots.org wrote:
Hi, all. I'm probably doing Something Dumb(tm), so please feel free
to
point out whatever I'm missing, no matter how stupid.
Anyway, I've got IAX set up to Vitelity. When I try to call my DID
Have a quick look at this guide on NAT and SIP -
http://www.aocomputing.net/?p=3. This is the link given if you were to ask
this same question in the IRC channel...
--wcs
On Wed, Oct 21, 2009 at 9:59 AM, Ott Rose sixfourimp...@hotmail.com wrote:
Here is what i think the is helpful from
of the atftpd instances)
/home/phones/cisco/7961 (root directory for another atftpd instance)
Then, with a little dhcpd.conf magic, you can easily point different sets of
phones to different tftp servers, using pools that match on key Product ID
(I think) strings.
--Warren Selby
On Fri, Oct 23, 2009
Have you tried accessing the IP address of your server from another
computer's web browser?
--Warren Selby
On Fri, Oct 23, 2009 at 10:19 AM, giancarlo lombardo
gianclomba...@gmail.com wrote:
Dear all,
I just installed asterixnow,
but no graphical interface start automaticaly neither linux
rough idea at the moment, but hopefully it gives you some ideas to work
with.
Thanks,
--Warren Selby
On Fri, Oct 30, 2009 at 12:35 PM, Danny Nicholas da...@debsinc.com wrote:
Have you tried “forward-porting” it? I don’t do queues or 1.6 so it’s
just an academic question to me
How are you setting up xlite and the ata? Which version of Asterisk are you
using? What does the general section of your sip.conf look like?
On Fri, Oct 30, 2009 at 1:01 PM, Cliconnect cliconn...@cliconnect.comwrote:
Hi all,
I can only get a line signal when I set the phones to not
You're attempting to connect on ports 5061-5062 but are bound to port
5060...?
What does your CLI look like during a failed call attempt?
Thanks,
--Warren Selby
On Fri, Oct 30, 2009 at 2:18 PM, Cliconnect cliconn...@cliconnect.comwrote:
Thank you,
How are you setting up xlite
Probably after 1.6.2 has been officially released beyond the release
candidate stage.
Thanks,
--Warren Selby
On Mon, Nov 2, 2009 at 4:14 AM, Khaled W Chehab kche...@xplorium.comwrote:
When we can expect to have a res_fax and res_fax_degium module for
asterisk V 1.6.2
Regards
What version of asterisk are you installing?
Thanks,
--Warren Selby
On Mon, Nov 2, 2009 at 5:59 AM, Dan Journo d...@keshercommunications.comwrote:
Hello,
Does anyone know where I can get an up to date guide on installing
CDR_MSQL?
VOIP-Info has old information.
Many thanks
Dan
,
--Warren Selby
On Nov 7, 2009, at 9:45 AM, Stephen Reese rsre...@gmail.com wrote:
On Sat, Nov 7, 2009 at 12:56 AM, Warren Selby
wcse...@selbytech.com wrote:
That typically means you've got an error in your phone specific
config file,
the SEP[MAC].cnf.xml.
You need to login to the phone via
,
--Warren Selby
On Nov 9, 2009, at 8:35 PM, Stephen Reese rsre...@gmail.com wrote:
On Sat, Nov 7, 2009 at 11:36 AM, Warren Selby
wcse...@selbytech.com wrote:
I think your featureLabel definition is wrong.
On the login issue, ssh to the ip of the phone and login first with
the user/pass you defined
://www.selbytech.com/2009/10/setup-cisco-7941-or-7961-with-asterisk/
In that post is a sanitized version of my conf file that I use on my
own deskphone, if you'd like to download it and try it out with your
setup.
Thanks,
--Warren Selby
On Nov 10, 2009, at 6:32 PM, Stephen Reese rsre...@gmail.com
already
linked to them in another post to the list).
Thanks,
--Warren Selby
On Thu, Nov 12, 2009 at 4:11 PM, Stephen Reese rsre...@gmail.com wrote:
On Wed, Nov 11, 2009 at 9:34 PM, Warren Selby wcse...@selbytech.com
wrote:
The 7960 and 79x2 use different sip firmwares and as far a I have seen
Which models of cisco phones (i.e 79x0, 79x1, 79x2, etc). And what do you
mean by VLAN issue.
Thanks,
--Warren Selby
On Sun, Nov 15, 2009 at 7:41 PM, Leif Madsen
leif.mad...@asteriskdocs.orgwrote:
Julian Lyndon-Smith wrote:
We have several types of phones, Cisco 79xx, Aastra 9133i etc. We
CLI output of calls that go through the local channel instead of the defined
channel would be useful to help diagnose what's going on here.
Thanks,
--Warren Selby
On Mon, Nov 16, 2009 at 4:01 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
Hi
I've been noticing an odd issue with our servers
voipprovider)?
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What does your provider see when you attempt to call them?
Thanks,
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On Nov 18, 2009, at 3:38 PM, Landy Landy landysacco...@yahoo.com
wrote:
Thanks for replying.
But how come I'm able to use a softphone to place calls from withing
the lan? I really dont get it. What ports
insecure=very
disallow=all
allow=ulaw
allow=alaw
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any typographical errors (which I mentioned what you
posted contained).
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There could be many reasons for this. You should show us the output of
your asterisk cli during a failed call attempt, and we can go from
there.
Thanks,
--Warren Selby
On Nov 20, 2009, at 5:23 PM, Brad Darr bd...@juniper.net wrote:
Hello,
I have been working on getting a Cisco 7961G
is the phone support.
Thanks,
--Warren Selby
On Nov 24, 2009, at 2:49 AM, Olivier oza-4...@myamail.com wrote:
Hello,
LLDP is more and more available on various network elements
(endpoint, switches, ...).
It seems to ease network configuration.
Do you have any experience with it ?
How
Do you have *11 registered in your voicemail.conf file? What does the
cli output look like when you try to leave a voicemail?
Thanks,
--Warren Selby
On Nov 28, 2009, at 7:22 PM, matthieu Nicaise techni...@thinkrosystem.com
wrote:
Hello everybody,
I'm using Asterisk ( 1.6.1.9
19 09 55
techni...@thinkrosystem.com commerc...@thinkrosystem.com
Thinkro System
http://www.thinkrosystem.com/
What is the output of 'ls -lh /var/spool/asterisk/voicemail/default/*11/' ?
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http
, since it's not appearing
your cli output. Make sure your extensions.conf file has been saved and
then try dialplan reload in the cli and then try calling extension *11
again.
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an error about not being able to find the config file and then the
phone will not boot up. Has anyone seen anything like this before?
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Why not do something with Background()? i.e
Playback(you-have)
SayNumber(${numMessages})
Playback(messages)
Background(press-1-or-2)
Then just be sure to record the audio files in the appropriate
directory...
Thanks,
--Warren Selby
On Dec 3, 2009, at 12:39 AM, Olivier oza-4
You need to purchase a smartnet license for the phone in question in
order to legally get the sip firmware.
Thanks,
--Warren Selby
On Dec 2, 2009, at 11:28 PM, Ricardo Melendez
rmelen...@utep.com.mx wrote:
Hi to All, I am trying to flash to SIP image one Cisco 7941 IP Phone
to work
:
No such command 'restart gracefully' (type 'help restart gracefully' for
other possible commands)
Can anyone think of why this is happening?
Thanks
Maybe you need to escape your quotes (\restart gracefully\) in your
script?
Just a thought...
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Can I install my free fax for asterisk license on more than one
machine? I.e using my digiun account to download the free FFA module,
am I restricted to just the first machine I put it on, or can I put
the free FFA on multiple servers?
Thanks,
--Warren Selby
On Fri, Dec 11, 2009 at 10:30 AM, Leif Madsen
leif.mad...@asteriskdocs.orgwrote:
Warren Selby wrote:
Can I install my free fax for asterisk license on more than one
machine? I.e using my digiun account to download the free FFA module,
am I restricted to just the first machine I put
Take the whitespace out of your ()'s. It's:
exten = 80,n,BackGround(es/good)
not
exten = 80,n,BackGround( es/good )
Thanks,
--Warren Selby
On Dec 12, 2009, at 9:16 PM, Landy Landy landysacco...@yahoo.com
wrote:
Same thing:
== Using SIP RTP CoS mark 5
-- Executing [...@outbound:1
the newest FFA modules will be released, or
if there is any way I can help to test the new modules?
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Is the new Fax For Asterisk being released in conjunction with this
release?
Thanks,
--Warren Selby
On Dec 18, 2009, at 4:59 PM, Asterisk Development Team asteriskt...@digium.com
wrote:
The Asterisk Development Team has announced the release of Asterisk
1.6.1.12.
This release
http://www.digium.com/en/products/software/faxforasterisk.php
Thanks,
--Warren Selby
On Dec 18, 2009, at 7:11 PM, Thomas Perron thomas.per...@gmail.com
wrote:
How does Fax for Asterisk work?
On Fri, Dec 18, 2009 at 7:51 PM, Leif Madsen
leif.mad...@asteriskdocs.org wrote:
Warren Selby
'.
Please run ./configure.
make: *** [makeopts] Error 1
And did you run ./configure like the error message says?
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And what is the output of the ./configure? Does it generate any errors?
Thanks,
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On Dec 22, 2009, at 1:09 AM, hadi motamedi motamed...@gmail.com wrote:
On Tue, Dec 22, 2009 at 6:56 AM, Warren Selby
wcse...@selbytech.com wrote:
On Mon, Dec 21, 2009 at 11:12 PM, hadi
://lists.digium.com/mailman/listinfo/asterisk-users
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are using, you can set the port
that asterisk binds to using the following commands in sip.conf:
1.6.x:
udpbindaddr = x.x.x.x:5061
1.4.x:
bindport = 5061
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Instead of host=dynamic, use host=1.1.1.1, or
host=1.1.1.0/255.255.255.0.
Thanks,
--Warren Selby
On Jan 12, 2010, at 11:16 AM, Aggio Alberto
alberto.ag...@loquendo.com wrote:
Hi guys,
I recently faced an issue regarding SIP registration: I have a 2-NIC
Linux PC, with eth0 set
the line, as well as allow you to follow several of the
simple how-to guides out there.
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set.
It's just not the easiest path to take, and not necessarily the path the OP
should go down unless he's looking for a challenge.
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as intended.
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are available for the Queue() command in the dialplan.
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was able to properly execute the command). If your script
doesn't properly handle these responses, you get the error mentioned
below.
It's never caused any of my calls to drop, though. Try turning on AGI
debug to see if this is the case for you.
Thanks,
--Warren Selby
On Jan 26, 2010, at 5:11
$ for someone just to
tell me how to configure it properly if it's a matter of just missing a
config line.
Mike
Which polycom phones are you using and what SIP firmware are you using?
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to offer an exact solution, but I'm very interested in
any results you may come up with.
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Be sure to set a timeout on your Queue() command in the dialplan.
exten = 100,1,Answer()
exten = 100,n,Queue(test1800)
exten = 100,n,Voicemail(100,u)
exten = 100,n,Hangup()
Thanks,
--Warren Selby
On Feb 8, 2010, at 3:27 AM, Louis-David Mitterrand
vindex+lists-asterisk-us...@apartia.org
I've
just missed something simple?
Thanks,
Warren Selby
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login attempts being a process
asterisk controls or just simply logs so that another tool can do the
banning, etc). I just don't remember if there was any followup to those
discussions.
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http://www.selbytech.com
the second peer up, and go from there?
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and the
conversation I had with the Cisco SPA rep at Astricon).
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wait time,
etc?
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with a '\'
[Syntax]
FILTER(allowed-chars,string)
[Arguments]
Not available
[See Also]
Not available
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sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
Or udpbindaddr for 1.6.2+...also, tcpbindaddr, tlsbindaddr if you plan
on adding TCP/TLS SIP support to asterisk.
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http://www.selbytech.com
will be connected with an answering machine and be requested to
leave name and phone number ;-)
Erik
One thing FILTER() will allow though is variable length dial strings, which
are needed in some parts of the world (as evidenced by earlier posts in this
thread).
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That's what I've started doing.
Thanks,
--Warren Selby
On Feb 17, 2010, at 8:29 AM, Miguel Molina mmol...@millenium.com.co
wrote:
Lenz Emilitri escribió:
Ok but this is available today and works fine, so it can be used as a
zero day replacement. Any syntax change is welcome
QUEUESRVLEVELPERF current service level performance
My version of asterisk is: 1.4.23.1
Thanks
ML
This doesn't directly answer your question, but you may look at setting a
unique filename before entering the queue, then do some post processing in
the h extension?
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On Wed, Feb 17, 2010 at 6:53 PM, Daniel Bareiro daniel-lis...@gmx.netwrote:
; DGB - 20100211
externip = sysadminhaiku.com.ar
localnet = 10.1.0.0/24
If you're using dynamic dns, shouldn't you be using externhost instead of
externip?
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file?
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version
You need the xen-specific kernel headers for your distro. If it's CentOS,
you would do it with yum install kernel-xen-header. I'm not sure on
Debian-based systems, but I think it's apt-get install kernel-xen-devel.
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', but the engine is not
available--
Is MySQL running and all the proper values set in the appropriate files?
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.conf file?
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You need to set your firewall public ip to dhcp in order for Uverse
dmz to work.
Thanks,
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On Mar 2, 2010, at 8:53 PM, sean darcy seandar...@gmail.com wrote:
Fred Posner wrote:
On Mar 2, 2010, at 6:27 PM, sean darcy wrote:
I've just got Uverse installed. I had dsl, but ATT
have a phone at home behind my RG, so I can't speak to your specific
situation.
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On Thu, Mar 4, 2010 at 1:00 PM, Dave Fullerton
dfullertaster...@shorelinecontainer.com wrote:
My Linksys PAP2T-NA at home has it's own clock / NTP settings that sends the
timestamp out to my analog phones. Check through the settings tab on your
Linksys for a time setting.
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phone for 20
seconds, then dial their cell phone for 40 seconds? Something like this:
exten = 100,1,Dial(SIP/100,20)
exten = 100,2,Dial(DAHDI/g1/${CELL_NUM},40)
exten = 100,3,Hangup()
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(${CALLINNUM},welcome)
Try setting CALLINNUM to ${EXTEN}.
exten = _95040X,1,Set(CALLINNUM=${EXTEN}).
This way you'll capture whatever is matching the _95040X pattern,
instead of the value in the CALLERID(dnid).
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is 100% wrong)
How is it wrong if he's creating his own variable named CALLINNUM?
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What version of asterisk are you using? Dialplan reload wasn't added
until 1.4. If for some reason you have a 1.2 or older asterisk
install, you'll need to use extensions reload (I think, I don't have
a 1.2 box in front of me to confirm the exact command).
Thanks,
--Warren Selby
On Mar
commands you can enter. You may need to do that and copy and
paste the output from that as well...
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New
to
voicemail and all the phones continued to ring. When this happens the
phones will continue to ring forever. The only way to stop them from
ringing is to pickup the handset at which time they realize there is no call
and reset.
What kind of phones?
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On Fri, Mar 12, 2010 at 4:25 AM, jonas kellens jonas.kell...@telenet.bewrote:
Are you using SIP realtime?
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the sources for the kernel
2.6.18-164.6.1.el5xen , so how can i find it?
http://wiki.centos.org/HowTos/I_need_the_Kernel_Source
yum install kernel-xen-devel.
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distclean before rerunning
./configure.
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.
Nope.
Oops. :) My mistake.
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On Mon, Mar 29, 2010 at 12:25 AM, Troy Davis t...@yort.com wrote:
sip fixup is enabled on the PIX
Try disabling the sip fixup on the PIX and see if that helps. You may have
to adjust the configs on the phones themselves when you do this.
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is the output of uname -i ? What commands did you use to install
your kernel-devel package. It looks like you need to do the following:
yum install kernel-PAE-devel
Is this what you're doing that's causing kernel conflicts?
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with the HL10 lifter and they're very happy
with them.
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Just some additional things for you to consider when building out your
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think in those cases,
however, it's a REGISTER request, not an INVITE. How is your sip.conf
configured for these end points?
Do you have any phones other than the ones experiencing this problem that
you can test with?
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from contacts though. You haven't
mentioned which softphone you're using, if you do that we may be able to
give you specifics for that softphone as well.
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at a client house or office and
they were connecting to a public server. Which version of the SIP firmware
are you using on your 7965?
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that their service
will recognize the private IP and rewrite the SIP packets. However
this is going to cause issues for my remote SIP phones.
Last I checked with Flowroute, they weren't yet supporting T.38. Has this
changed in the last month or so?
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http
have any
advice on what might be causing this and where the first place I need
to look is?
Thank You,
--
Steve Anness
I've seen this sometimes and the quick and dirty solution was to restart
asterisk. Check core show channels and see if there are any 'hung'
channels.
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directory `/usr/src/kernels/2.6.18-164.11.1.el5-i686'
make: *** [modules] Error 2
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.
Thanks.
What is UUI?
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--Warren Selby
http://www.selbytech.com
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headers:
]# yum remove kernel-devel
]# yum install kernel-devel
Thanks,
--Warren Selby
On May 25, 2010, at 9:48 PM, Warren Selby wcse...@selbytech.com wrote:
I was at a client site tonight to install OSLEC on his machine
running asterisk 1.6.0.22 and DAHDI 2.2.1 installed via yum. I
stopped
, but you should be able to find it if you read through
that file.
Thanks,
--Warren Selby--
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passwords.
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Thanks,
--Warren Selby
http://www.selbytech.com
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http
.
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Thanks,
--Warren Selby
http://www.selbytech.com
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On Fri, Jun 18, 2010 at 7:44 AM, Aksel Celasun ak...@abacus-it.no wrote:
Minor edit on the include = nighttime|12:30-8:00|mon-fri|*|*
Correct now.
This isn't how you do time based checks in asterisk. Lookup the application
GotoIfTime.
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Thanks,
--Warren Selby
http://www.selbytech.com
a config file called
dialplan.xml. I think on the SPA5xx series you can configure this parameter
either in the main config file or from the web interface. You need to look
for something like dialplan or dial plans, etc. This controls the
timeout when entering digits.
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Thanks,
--Warren Selby
.
The split contains all the firmwares for the different model phones as
separate files, the combined combines all of the firmwares into one big
firmware file. The combined will cover any supported polycom phone model,
but it takes longer to load.
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Thanks,
--Warren Selby
http://www.selbytech.com
that works with a 7941 on my website (you can find the link my
signature), I think with a little adaptation you can make it work with a
7965.
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--Warren Selby
http://www.selbytech.com
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. It will take a little trial and error, but you should be able
to get it done.
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Thanks,
--Warren Selby
http://www.selbytech.com
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that I'm trying to grab a date, but it's not taking the date format
parameters that I want.
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Thanks,
--Warren Selby
http://www.selbytech.com
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New
. I'd rather they be set in the [globals]
context if at all possible, that way I'm not limited to where I can use
them.
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--Warren Selby
http://www.selbytech.com
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