this working. It also helps that we're
going VoIP at work.
Once I demonstrate to my wife how cool this is, (will be?) I intend to get a
telephony board and do Really Cool Things. (tm)
Thanx,
Mike Diehl.
On Friday 21 March 2003 12:09 am, WipeOut . wrote:
Yes it is as good as it sounds
On Friday 21 March 2003 1:49 pm, Jeremy McNamara wrote:
Mike Diehl wrote:
Unfortunately, I'm kinda commited to NetMeeting. The problem is that I
have friends with a mix of Windows and Linux machines. Between
Netmeeting and Gnomemeeting, I should be able to get everyone connected.
Good guy
(Critical Response)
[Sep 23 08:40:21] WARNING[21450] chan_sip.c: Hanging up call [EMAIL PROTECTED]
- no reply to our critical packet.
===
Am I reading and understanding these log entries correctly?
Thank you for your help,
--
Mike Diehl
On Sunday 23 September 2007 06:43:54 pm Paul wrote:
Mike Diehl wrote:
I just had a user complain about a call getting dropped and another one
failing to go through.
I'm trying to interpret the log entries for each call and would like to
confirm my understanding.
The first entry is from
and do the right thing.
What am I missing?
TIA,
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Shameless plug:
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tried to use the get_option() method that was documented in the
module POD file; Perl complains that the method isn't defined:
$result = $agi-get_option($msg, 12345, 1);
So, what am I missing? I know this works; too many people are doing it. Any
ideas?
TIA,
--
Mike Diehl
worked as expected.
Thank you for your time.
Mike.
On Wednesday 23 July 2008 01:48:14 pm David Van Ginneken wrote:
Mike Diehl wrote:
Hi all,
I'm trying to build an IVR using the Perl AGI module at
http://search.cpan.org/~jamesgol/asterisk-perl-0.10/lib/Asterisk/AGI.pm
But, I'm having
On Thursday 24 July 2008 12:58:29 am Steve Edwards wrote:
The agi debug command (1.2) would have shown you where you violated the
protocol.
Nice to know...
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fails, my friend does hear the phone ring. BTW, I'm running
Asterisk 1.4.4.
Does anyone know how to fix this?
TIA,
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On Thursday 24 May 2007 06:35, Steve Murphy wrote:
On Wed, 2007-05-23 at 20:51 -0600, Mike Diehl wrote:
Hi all,
I'm having a problem with an asterisk server being unable to call certain
cellphones and answering machines. Anytime the person answers the phone
call, everything works well
can look for improvement?
TIA,
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]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike Diehl
Sent: Tuesday, June 19, 2007 12:27 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Need to increase call count
Hi all.
I've got a project where I need to make outbound calls and play a
prerecorded .wav file to the called
://www.diehlnet.com/features.conf
Any ideas are welcome.
TIA,
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Actually, it turns out that sometimes I can't get ANY DTMF to work. I can
call a local phone number and log into my voicemail system at work. But my
wife is unable to dial a toll free number and use their IVR. Hope this
helps.
On Wednesday 28 March 2007 16:58, Mike Diehl wrote:
Hi all
to perform special actions, maybe thats your problem. also
check for your dtmf setting. dtmf settings should be same on both sides.
On 3/29/07, Mike Diehl [EMAIL PROTECTED] wrote:
Actually, it turns out that sometimes I can't get ANY DTMF to work. I
can call a local phone number and log into my
No, this is just a standard phone service.
On Sunday 01 April 2007 20:43, Steve Totaro wrote:
Is this while using queues?
Mike Diehl wrote:
Hi all,
I've trying to fix a problem. If I'm in a call and I press the * key,
the call goes silent but doesn't hang up. I need to be able
and a Polycom 501. Any
ideas where to start?
Thanx,
Mike Diehl.
On Thursday 29 March 2007 11:52, Doug wrote:
At 18:23 3/28/2007, Mike Diehl wrote:
Actually, it turns out that sometimes I can't get ANY DTMF to work. I
can call a local phone number and log into my voicemail system at work
on the console. Sounds like it's a phone issue
after all, right?
I've got the same symptoms for BOTH the Sipura 2002 and a Polycom 501.
Any ideas where to start?
What is your dtmfmode set to in sip.conf (e.g. inband, rfc2833)?
- Noah
Thanx,
Mike Diehl.
On Thursday 29 March 2007 11:52
other DTMF
presses too.
Regards,
Steve
On 4/4/07, Mike Diehl [EMAIL PROTECTED] wrote:
There wasn't a setting, but I set it to rfc2833.
On Wednesday 04 April 2007 12:49, Noah Miller wrote:
Hi Mike -
Well, when I restart the cli as requested below and go the addition
steps
Hi all,
I'm looking into being able to send/receive SMS messages with my
asterisk box in the US. I've seen the SMS command as well as the Kannel
program. I'd prefer to do it from Asterisk.
I've tried something like:
exten = 999,n,sms(15551234567,s,This is a test)
in my dialplan, but when
On Fri, 2009-02-20 at 13:46 +1300, Michael wrote:
I also think you should check the economic stimulus package. There are
billions of dollars allocated to ISPs. It could be a windfall.
Yoohoo! Let's print some more money. I don't think that's been tried
before
Of course it has!
Hi all,
I've got a problem where many times, there is silence at the first 1-2
seconds of a call. Then it clears up and it's crystal clear. I've not
put a sniffer on it, yet, but I suspect that the media channel is still
being set up. The server shouldn't be too overloaded. Can anyone give
me
speaking. For example, when I answer a call, I don't say, Hello until
I hear a bit of noise on the channel which takes a second. If it's
longer than about a second maybe you have some other issues to deal with.
Mike Diehl wrote:
Hi all,
I've got a problem where many times, there is silence
Hi all.
I received a PAP2T-NA from a potential customer to see if I could get it
configured for testing. I plugged it into my network and plugged a phone
into it and attempted to do a factory reset from the handset.
I pressed and got NOTHING! Just silence. So, is this TA a brick? Or
it installed somewhere in production). Sniff the
packets coming out of it to see if you can determine its IP, but I am
guessing if the previous owner already disabled the IVR, they probably
locked down the device pretty well :) Are you sure it is an NA?
j
On Mon, 30 Mar 2009, Mike Diehl wrote:
Hi
Hi all.
As a convieneince to my users, I'm trying to strip off the leading 1 and
areacode from incoming calls. However, when I do, the src field in the CDR
is also stripped. I'd like the CDR to reflect the connonical form of the
incoming number.
Any way do to this?
TIA,
Mike Diehl
of the alpha-page?
TIA,
Mike Diehl.
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Yup, that did it. Thank you.
On Tuesday 26 September 2006 23:23, Lacy Moore - Aspendora wrote:
Look for pagerbody and pagersubject.
On 9/26/06, Mike Diehl [EMAIL PROTECTED] wrote:
Hi all.
I currently get an alpha-page via email from Asterisk when I get a new
voicemail message
sipsock_read: SIP MESSAGE JUST
IGNORED: BYE
Oct 18 21:18:32 ERROR[22095]: chan_sip.c:11325 sipsock_read: BAD! BAD! BAD!
==
I'm running Asterisk 1.2.9.1 . Any ideas what this means, and should I be
concerned?
TIA,
Mike Diehl
happen to know that November's Linux Journal will have an article about
running Linux/Asterisk on a Linsys WRTGS54SL router. Nothing too
technical, but I hope you enjoy it.
Mike Diehl.
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Doh! Turns out it won't be November. It will be a bit later. Sorry.
On Thursday 19 October 2006 21:35, Mike Diehl wrote:
On Thursday 19 October 2006 14:10, Cory Andrews wrote:
I caught a thread the other day concerning Astricon and users embedding
Asterisk on a Linksys or Netgear
I don't know how big your voicemail system is, but have you considered using
Unison to syncronize the vm accross all your servers? I'm deploying multiple
servers with two vm servers, each sync'ed every 5? minutes. If one fails,
the other one should be good enough.
Just a though,
Mike
On
On Saturday 17 June 2006 01:55, Tzafrir Cohen wrote:
On Fri, Jun 16, 2006 at 09:40:35AM -0600, Mike Diehl wrote:
I don't know how big your voicemail system is, but have you considered
using Unison to syncronize the vm accross all your servers? I'm
deploying multiple servers with two vm
On Saturday 17 June 2006 10:03, Douglas Garstang wrote:
Yes, we'd need it on every single box. We had a dedicated voicemail server
in the first place. I decided to distribute voicemail between all boxes
because the script that I had that copied the phone registrations over to
the voicemail
Hi all.
We are in the process of doing some VoIP testing with the intent to eventually
replace our 5ESS phone switch.
However, during the transition period, we'd like to be able to use our
existing voicemail system which is Octel. It's pretty easy to figure out how
to change the dialplan to
that functionality.
BTW, the Octel is connected to the 5ESS via T1, as will the Asterisk server.
Hope this helps you help me.
Thank you,
Mike.
On Wednesday 26 July 2006 10:57, Olivier wrote:
2006/7/26, Mike Diehl [EMAIL PROTECTED]:
We have ISDN phones that have a Message Light that we don't want
the CID number, it changes the
CDR(src) field.
Is there any way I can have the best of both worlds?
TIA,
Mike Diehl.
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was sure hoping there was a better way.
Thanx for your time.
Mike Diehl.
On Sunday 29 October 2006 23:01, Leo Ann Boon wrote:
Mike Diehl wrote:
Hi all,
I've been beating my head against this for some time now.
For incoming calls, I'd like to send my users a localized caller id
number
suspect it has something to do with the features.conf file, which you can
look at at:
http://diehlnet.com/features.conf
Otherwise, any advise would be most welcome.
Mike Diehl.
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something?
Thanks in advance,
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of the context that you want to
use.
That may work for you.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
Sent: Saturday, 10 October 2009 6:28 AM
To: Asterisk Users Mailing List - Non-Commercial
to fix it?
TIA,
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sanitized, of course.
I'm starting to get deparate to fix this...
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Mike Diehl
On Saturday 28 November 2009 06:54:16 am Kevin P. Fleming wrote:
Mike Diehl wrote:
On Friday 27 November 2009 11:09:02 am Noah Miller wrote:
Hi Mike -
I've got a Polycom 501 that's been working with Asterisk for some time.
However, I don't seem to be able to put a call on hold and get
On Saturday 28 November 2009 04:48:13 pm Darrick Hartman wrote:
Mike Diehl wrote:
The phone is a Polycom 501; it's been discontinued. I am working on a
testing/migration plan to move to the latest Asterisk 1.6.x, but I'm
hesitant to upgrade a system that doesn't currently work right
On Saturday 28 November 2009 04:48:13 pm Darrick Hartman wrote:
Mike Diehl wrote:
On Saturday 28 November 2009 06:54:16 am Kevin P. Fleming wrote:
Mike Diehl wrote:
On Friday 27 November 2009 11:09:02 am Noah Miller wrote:
Hi Mike -
I've got a Polycom 501 that's been working
as I can.
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On Saturday 28 November 2009 06:48:01 pm Darrick Hartman wrote:
Mike Diehl wrote:
On Saturday 28 November 2009 04:48:13 pm Darrick Hartman wrote:
Mike Diehl wrote:
On Saturday 28 November 2009 06:54:16 am Kevin P. Fleming wrote:
Mike Diehl wrote:
On Friday 27 November 2009 11:09:02 am
/diehlnet.txt
http://www.diehlnet.com/Polycom-0004f211d1d0.txt
I've changed the extensions on the website from .cfg to .txt so that it will
open better for you.
What have I done wrong?
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call. Can this be done?
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configuration for dnsmasq?
If I can't get this working, I'll have to resort to hard-coding the
information into each of 12 phones Yuck!
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before? Any clues as to how to fix it?
TIA,
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Mike Diehl.
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this happen once, but I've been unable to reproduce it reliably.
Any ideas?
Mike Diehl.
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Hi all,
I'm trying to get moh working on * version 1.4.4. I've setup a test
extension that answers the call and runs the musiconhold command with
the appropriate class name.
All I get on the phone is silence. The console tells me that moh
started and immediately stopped, but it complains
David Backeberg wrote:
On Mon, Feb 22, 2010 at 2:20 PM, Mike Diehl mdi...@diehlnet.com wrote:
Hi all,
I'm trying to get moh working on * version 1.4.4. I've setup a test
I don't know the answer, but are you really using 1.4.4? If so,
consider taking some time to review the security
done.
Let me know if you need more help.
I owe you one, btw, because I read your blog on getting these beasts
provisioned in the first place. sip:15058228...@robodial12.diehlnet.com
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{
hardware ethernet 00:04:F2:27:8F:F8;
}
host 0004f22afafd{
hardware ethernet 00:04:F2:2A:5A:FD;
}
}
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?
Otherwise, is there a product/service they can buy that will allow them to fax
to/from their computers?
TIA,
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Mike Diehl.
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On Monday 29 March 2010 10:15:50 am jon pounder wrote:
Mike Diehl wrote:
Hi all,
I've cross-posted this to the -users and -biz groups. Hope that's OK.
I have a customer who REALLY needs to be able to send/receive faxes
reliably. I could probably get hylafax configured, but I'm
?
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have to configure.
BR
- Andrea
In the database, I changed the accountname to test and tried:
sip show peer test load
Asterisk replied:
Peer test not found.
So it looks like I'm missing something pretty basic.
Any ideas?
--
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Mike Diehl
the server is idle.
I've got allow = all in my sip.conf file.
Anyway, does anyone have an idea on how to resolve this?
--
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On Monday 11 October 2010 4:34:46 am Stefan Tichy wrote:
On Sun, Oct 10, 2010 at 06:29:27PM -0600, Mike Diehl wrote:
Asterisk replied:
Peer test not found.
So it looks like I'm missing something pretty basic.
I would suggest to check extconfig.conf.
That's where the problem was; I
Never mind...
I mistakenly interpreted codec_a_mu.so as some sort of universal translator
between ulaw, alaw, and slin. When I loaded the rest of the modules, it
worked like a champ.
Mike.
On Sunday 10 October 2010 6:57:30 pm Mike Diehl wrote:
I'm doing some final check-outs before
limitation, or am I doing something wrong? If this won't
work, is there a work-around?
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Mike Diehl.
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be able to get it working from here.
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Mike Diehl.
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simply aren't being sent.
It only seems to happen on a particular trunk. The same phone calling on a
different trunk works just fine.
Any ideas?
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forwarding settings, and check that the RTP ports that
have been negotiated for the call are not conflicting with those of other
devices/calls/port forwarding settings.
Philipp
--
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Mike Diehl
Hi all,
I've got two questions about CDR's.
1. I'd like to start logging the IP address that a call orginates from.
I'm sure I can get this into the userfield of my CDR table, but what
variable should I use to get this value? I looked at the variables page at
voip-info and didn't find
minutes. We'd like to be able to start our message as soon as the greeting
is done.
Any suggestions?
TIA,
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New
and replace all of the
office phones.
With these short distances, will I need to worry about echo? Do these
devices have echo cancellation?
TIA,
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, as far as I can
see.
Is there a way to get this information from the directory application?
TIA,
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Mike Diehl.
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Steve Edwards asterisk@sedwards.com wrote:
On Mon, 31 Jan 2011, Mike Diehl wrote:
I've got an agi script that calls the directory function, which seems to
work to a point. However, once the caller has selected an entry, I need
my agi script to find out which extension was selected
configured codec is u711.
When the user tries to send a fax, it gets to the point where it issues a
reInvite to start the T.38, then the called side receives a SIP 488 (Not
Acceptable Here)
Where should I start? Any pointers would be most welcome.
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to Asterisk? Join us for a live introductory webinar every Thurs:
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message on the console, I've set verbose to
25. Any ideas? I'd like to take the next few instructions to log
success/failure.
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; to indicate a failed transfer
However, it seems that transfer is a function of the phone, not Asterisk. Is
there any way I can configure the Polycom phones to either use the Asterisk
function, or to make a beep when a transfer completes?
TIA,
--
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Mike Diehl
Hi all,
Is it possible to send a SIP header to a PAP2T or SPA and cause the device
to automatically answer? I can do this with my Polycom phones and would like
to do it with my ATA's.
Any ideas?
--
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) that is powered by an
analog line and can do auto answer when it gets the first ring.
I'll look into it. Thank you.
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New
Hi all,
Is it possible to store voicemail in a Mysql database without using ODBC?
I've got RTA sip and voicemail working; I just want to store the messages in
the db now. Configuring ODBC seems like a lot of work if I don't have to.
TIA.
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Mike Diehl
. The sip database records are virtually identical, btw.
Any ideas?
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are in place because my Polycom and Grandstream servers work
just fine.
What else do I need to do to get the PAP to work this way?
TIA,
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am I missing?
Mike.
On Tuesday 14 June 2011 3:08:33 am Paul Hayes wrote:
On 13/06/11 19:44, Mike Diehl wrote:
Hi all,
I'm trying to provision my PAP2T's to use a SVR lookup to find the
Asterisk server. I'm using a provisioning file that contains an element
like:
Proxy_1_ _sip
? Should I
move to 1.8, yet?
So far, the only work-around I've come up with is a separate context for EACH
sip account, with the variables hard-wired FUGLY!
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On Thursday 16 June 2011 8:10:38 am Marius Pedersen wrote:
On 06/16/2011 07:58 AM, Mike Diehl wrote:
Well, I ran a simple test by trying to configure the second port to use
the DNS SRV record, as described below.
Here is what I have: (sanitized
are registered.
0004F2127F60-1 is the registration name for line 1 and 0004F2127F60-3 is for
line 3.
Any ideas on how to fix this? Would doing a factory reset and
reprovisioning on the phone help? Or would that be just wheelspin?
TIA,
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scheduling a SIP reload tonight. Might as well do an Asterisk restart
instead. grin
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Hi all,
I've got a number of Grandstream phones and I'd like to be able to reboot them
remotely, as I do my Polycoms...
I've got this in my sip_notify.cfg:
[grandstream-check-cfg]
Event=sys-control
Doesn't seem to work. Any ideas?
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Mike Diehl
Hi all,
I'm trying to figure out how it is that a couple lines on a given phone, with 3
lines, can qualify as unavailable while the remaining lines can be available.
I've got qualify=1000 in my sip.cfg.
Shouldn't this be an all-or-nothing proposition?
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Mike Diehl
some of the phone lines go down and they stay down until the phone is
rebooted.
I'm not even sure what to look for when I go to the site. Any ideas?
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-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Mike Diehl
Sent: Friday, 22 July 2011 10:50 a.m.
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Rebooting a Grandstream
Hi all,
I've got a number
On Friday 22 July 2011 1:42:33 am Ishfaq Malik wrote:
On Thu, 2011-07-21 at 17:13 -0600, Mike Diehl wrote:
Hi all,
I've got a strange problem with a customer's phones.
They've got a bunch of Grandstreams that seem to be rock solid... until
7:00pm. At 7:00, some of the phones become
][501][962522A][0101062C] 000B821CA9B6-1 SIP registration
failed.
Retrying in 20 seconds. Server: 209.250.31.96
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On Friday 22 July 2011 2:38:15 am Mike Diehl wrote:
On Friday 22 July 2011 1:42:33 am Ishfaq
running against one of the phones on my server, but so far,
it's not rebooted, so I've got nothing to look at.
Any other ideas?
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Take care and have fun,
Mike Diehl.
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On Wednesday 17 August 2011 4:12:21 pm Doug Lytle wrote:
Mike Diehl wrote:
Any other ideas?
They should be writing out logs to your ftp server (If your provisioning
them that way).
At the moment, my web server isn't capable of receiving the phones POST
request. Sounds like that's going
On Wednesday 17 August 2011 4:11:32 pm Andrew Latham wrote:
On Wed, Aug 17, 2011 at 6:01 PM, Mike Diehl mdi...@diehlnet.com wrote:
Hi all,
I've got a customer with 10 Polycom 335's and the latest(ish) firmware.
For the most part, things are working well.
However, about once a day
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