in "$@"; do
> if [ -d ${BASEDIR}${ext} ];then
>for msgdir in $(ls -d ${BASEDIR}${ext}/*); do
> ProcessDir ${msgdir}
>done
> else
>echo "${BASEDIR}${ext} is not a valid directory"
> fi
> echo "Processed extension $ext&qu
mailbox:
> https://github.com/asterisk/asterisk/issues/181
>
> Hope this helps.
>
> BR,
> -Mike
>
> On Mon, Oct 9, 2023 at 1:06 PM Mike Diehl wrote:
> > Hi all,
> >
> > I need to be able to delete a voicemail message using a program.
> >
> >
Hi all,
I need to be able to delete a voicemail message using a program.
Is is sufficient to simply delete the .wav and .txt files in the spool
directory?
Or do I need to also renumber the remaining files?
For example, let say a given mailbox has 20 messages in it and I want to
delete mess
as to where I should start looking?
Thanks in advance,
--
Mike Diehl
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.o
On Thursday, May 16, 2019 05:12:17 PM Joshua C. Colp wrote:
> On Thu, May 16, 2019, at 5:00 PM, Mike Diehl wrote:
> > Hi all,
> >
> >
> > I've got a program that connects via AMI and acts upon the voicemail
> > message waiting event.
> >
> >
&g
d?
Thanks
--
Mike Diehl
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
ehind a NAT router and my server is on the open Internet.
>
> Calls within their office sound fine. Calls to/from most numbers sound
> fine.
>
> When they took their phones home, those same phone numbers still had
> problems.
>
> So, I don't think it's their net
My comments below:
On Wednesday, March 20, 2019 12:19:08 AM Antony Stone wrote:
> On Tuesday 19 March 2019 at 21:36:53, Mike Diehl wrote:
> > Hi all,
> >
> > I have a user who is reporting one-way audio, but only when a call is made
> > to or from particular PS
work correctly.
Any ideas where to look to fix this?
Thanks in advance.
--
Mike Diehl
--
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g wrong? Or is this approach simply doomed?
Any thoughts would be welcome.
--
Mike Diehl
--
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Well, it SEEMS to be working now. I don't know what I did, and frankly,
don't have time to back track to find out.
Thanks for your time.
Mike.
On Thu, May 24, 2018 at 4:33 AM, Doug Lytle wrote:
> On 05/23/2018 05:23 PM, Mike Diehl wrote:
>
>
> However, my user isn&#x
Hi all,
I've got an AGI script that launches the conference bridge with a line like:
"$main::agi->exec(ConfBridge,$conf,default_bridge,default_user,$menu_profile)"
The $conf variable contains the room number.
I'm trying to configure it so that when only one person is in the
conference, they hea
Hi all,
I have a user who would like to stream their favorite radio station from
iHeart radio for their music on hold.
It this TECHNICALLY possible? If so, any pointers would be appreciated.
Is this LEGAL in the US?
Thanks in advance,
Mike.
--
il Record (CDR) settings
--
Logging:Enabled
Mode: Simple
Log unanswered calls: No
Log congestion: No
* Registered Backends
---
cdr-custom
Adaptive ODBC
Any ideas would be appreciat
il Record (CDR) settings
--
Logging:Enabled
Mode: Simple
Log unanswered calls: No
Log congestion: No
* Registered Backends
---
cdr-custom
Adaptive ODBC
Any ideas would be apprec
If you'll release it for python, I'll take a stab at porting it to perl.
Mike
On October 19, 2017 4:53:52 PM EDT, Jonathan H wrote:
>That's because it uses a deprecated API and endpoint.
>
>However, funny you should ask this, because I've just finished
>updating my Google TTS routine to take ad
ike the idea of the IP geo location firewall rules coupled
with the "friendly scanner" filter, as provided by a few of you guys. It
was mentioned that this is a broad hammer, but I'm kinda looking for a
broad hammer! ;^)
Looks like I need to do some research, but I think I have what I need.
Man, I was hoping it was something like that. I did read the release notes; I
must have missed that part.
This should solve the problem, so thanks again.
Mike
On July 20, 2017 1:09:08 PM EDT, Richard Mudgett wrote:
>On Thu, Jul 20, 2017 at 11:50 AM, mdiehl
>wrote:
>
>> I recently upgraded As
in.
I'm running Asterisk 13.14.0 built by root @ server on a x86_64 running Linux
on
2017-06-20 14:27:06 UTC
For odbc, I've got unixODBC 2.3.2-r2.
Are these the versions I should be using? If so, any recommendations as to how
to
troubleshoot this would be
I could find out what syscall was being interrupted That MIGHT
tell me what was wrong, but this is all I get from strace.
Any ideas would be welcome.
Mike.
On Wednesday, June 07, 2017 04:34:10 PM Mike Diehl wrote:
> Thank you for your time. I've put my replies to your questions in-li
Thank you for your time. I've put my replies to your questions in-line, below.
On Wednesday, June 07, 2017 10:19:41 AM Antony Stone wrote:
> On Tuesday 06 June 2017 17:54:59 Mike Diehl wrote:
>
> > Hi all,
> >
> > I'm upgrading to Asterisk 13.14.0 x86
7-r1
I suspect that the odbc drivers are the problem. Is ther an alternative drive
that I should be using?
Failing that, any other ideas?
Thanks in advance.
--
Mike Diehl
--
_
-- Bandwidth and Colocation Provid
on my part.
Hope this helps someone else.
Mike.
On Thursday, April 06, 2017 10:28:03 AM you wrote:
> On Thu, Apr 6, 2017 at 10:20 AM, Mike Diehl wrote:
> > I found it!
> >
> > I had customized the safe_asterisk script and managed to slip in a -c on
the asterisk command li
, if
> realtime, what type of database, what channel drivers (SIP or PJSIP,
> and others).
>
> Matthew Fredrickson
>
> On Fri, Mar 31, 2017 at 12:08 PM, Mike Diehl wrote:
> > Hi all,
> >
> > I've upgraded to Asterisk 13.14.0 and now I'm seeing that
es; nothing resolved the issue.
Any suggestions?
--
Mike Diehl
--
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New to
produce and I can't emulate easily.
Any recommendations would be very welcome.
--
Mike Diehl
--
_
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produce and I can't emulate easily.
Any recommendations would be very welcome.
--
Mike Diehl
--
_
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Hi all,
I've got a device that seems to become unreachable for about 2 minutes, every
hour. From what I can tell, it isn't due to network or server issues. Any
ideas?
TIA.
--
Mike Diehl
Diehlnet Communications, LLC.
Voice: (505) 903-5700
Fax: (505
bridge to be created?
Thanks again,
Mike.
On Saturday, April 16, 2016 04:18:44 PM Bobby Hakimi wrote:
> You can't see them until someone joins the bridge, might be able to put in
> db using the asterisk live setup
>
> On Apr 16, 2016 1:36 PM, "Mike Diehl" wrote:
>
le structure do I use?
Thanks in advance,
--
Mike Diehl
Diehlnet Communications, LLC.
Voice: (505) 903-5700
Fax: (505) 903-5701
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join u
ax; check the manual that corresponds
>
> On Mar 23, 2016 11:38 PM, "Mike Diehl" wrote:
> > Hi all,
> >
> > I've got a new server up, but it's not staying up
> >
> > After a day or so, it segfaults with:
> >
> > [Mar 22 23:
GI script looks, essentially, like:
$main::agi->exec("ConfBridge","1505xxx");
I've got a dummy /etc/asterisk/confbridge.conf file:
[general]
[default_bridge]
type=bridge
[default_user]
type=user
[default_bridge]
type=bridge
[1505xxx]
type=bridge
Any su
that is stable. In
the mean time, I'm trying to run unixODBC 2.3.2.
What version SHOULD I use?
TIA,
--
Mike Diehl
Diehlnet Communications, LLC.
Voice: (505) 903-5700
Fax: (505) 903-5701
--
_
-- Bandwidth and Colocatio
Hi all,
I've got a couple of SPA112's that are having problems registering line 2.
Line 1 registers just fine. All of them are behind a NAT.
Here is a sample provisioning file that the devices are using.
(Any help would be most appreciated.)
Yes
xxx
syslog.example.com
3
Yes
Yes
2
360
isk versions and their supported lifetimes,
> please see the following wiki page:
>
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
>
> Thank you for your continued support of Asterisk!
Is there any time frame for when FFA will be available for 13?
--
Mike Dieh
On Tuesday, August 05, 2014 05:19:55 PM Steven Howes wrote:
On 5 Aug 2014, at 17:10, Mike Diehl wrote:
All of my SPA112's are running 1.3.2(014). My SPA8000's are running 5.1.10.
If you do firmware upgrade your 8000s, don’t go past 6.1.3 or it’ll go badly…
Freezing and requi
f my SPA112's are running 1.3.2(014). My SPA8000's are running 5.1.10.
--
Mike Diehl
Diehlnet Communications, LLC.
Voice: (505) 903-5700
Fax: (505) 903-5701
--
_
-- Bandwidth and Colocation Provided by http://www.api-dig
Hi all,
I've got a few devices, SPA112's and SPA8000's, that are giving me problems.
Each device has a separate SIP credential for each port, but sometimes, only a
few of the ports register.
So, the device will be running fine for a while, then suddenly one or more of
the ports will become U
Hi all,
I have a user with an old Mitel PBX connected to a couple of SPA112's. The
user is reporting that their phones ring several times a day and when they
answer the "call," all they hear is dial tone or busy signal.
Their PBX guy says that the SPA112's aren't providing line supervision and
t
Hi again,
I've got a user who's using a bunch of Grandstream GXP2xxx's. For the most
part, they work, except for when they try to do a phone-based call transfer.
Here's what it looks like is happening:
Phone A is on a call with phone B. (B could be another phone, or a PSTN
endpoint.)
The use
amp; tags).
>
>
>
> Also, have you verified with logging on the provisioning server that the
> configuration file is actually being pulled?
>
>
>
>
>
> Thank you,
>
>
>
> Noah Engelberth
>
> MetaLINK Technologies
>
>
>
> *From:* asteri
Hi all,
I've got a provisioning file that I use to configure Cisco SPA112's.
I'm wanting to get this file to do 3 things for me. I want to turn T.38
on, Call forwarding off, and Call waiting, off for both lines. but it's
not working.
This is what I'm using to enable T.38 for line 1.
Yes
1
Yes
Hi all,
I have a user who is reporting dropped calls at his site. We don't have
any other users complaining of this.
So far, this is what we know:
1. The manager bought all new Polycom phones. (POE)
2. They replaced the network switch with a POE version.
3. It's not just one or two of the
Hi all,
I'm installing Hylafax on my Asterisk system. From what I've read, I can
either use IAXModem or T38Modem to provide the virtual fax device. So at
the risk of starting a religious war, which one should I use?
I don't mind running IAX if I have to. I want as much flexibility and
stabilit
fax machine
>> is resending the document multiple times!
>>
>> Mike.
>>
>>
>> On Mon, Mar 10, 2014 at 12:49 PM, Steve Underwood > ste...@coppice.org>> wrote:
>>
>> On 03/11/2014 12:36 AM, Mike Diehl wrote:
>>
>> Hi all,
t; On 03/11/2014 12:36 AM, Mike Diehl wrote:
>
>> Hi all,
>>
>> For the most part, we are finding that Fax for Asterisk works pretty
>> well. However, we have seen some wierdness that we'd like to try to
>> fix.
>>
>> Once in a while, we will get a
Hi all,
For the most part, we are finding that Fax for Asterisk works pretty
well. However, we have seen some wierdness that we'd like to try to
fix.
Once in a while, we will get a partial result report for a given fax
but when we look at the actual .tiff image, it looks to be complete.
This is
party to answer before completing the transfer?
>
> --Don
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
> Sent: Monday, February 24, 2014 12:24 PM
> To: Asterisk
Hi all,
I have a user who is having trouble transferring calls, using a
Grandstream GXP2xxx.
Here's the use case that I've seen:
I call the user from phone A and he answers on phone B.
Then, he hits the transfer button on his phone and dials an extension
that is reachable by him, but not by me,
the fax, but not
the final delivery.
Does that make more sense?
Mike.
On Wed, Feb 19, 2014 at 6:10 PM, Matthew Jordan wrote:
> On Tue, Feb 18, 2014 at 2:13 PM, Steve Edwards
> wrote:
>> On Mon, 17 Feb 2014, Mike Diehl wrote:
>>
>>> Is there something I need to do in
Hi all,
I'm trying to build a fax relay mechanism where faxes come in and get
relayed out to their final destination. I'm using the h extension to store
various results from both legs. This data is being saved correctly for the
first (receiving) leg. The second leg isn't calling the h extension
Hi all,
I've got a customer who's reporting "ghost calls." Essentially, the phone
rings, they pick up, and there's no body there.
It is NOT one-way audio, and it doesn't happen all the time.
We use voipmonitor to watch calls, and this is what we saw for the call in
question:
| calldate
Based on what we're hearing, we've decided to replace the SPA112. Thank
you for your input.
Mike.
On Thu, Feb 6, 2014 at 4:39 PM, Andres wrote:
> On 2/6/14, 11:18 AM, Mike Diehl wrote:
>
> Hi all,
>
> I have an SPA112 that in sitting behind a Ubee cable modem
t;
> http://401stblow.wordpress.com/2012/10/21/fixing-time-warner-cable-ubee-modem-connectivity-issues/
>
>
> On Thursday, February 6, 2014, Mike Diehl wrote:
>
>> I've got the registration period set to 15 minutes. However, I've got
>> similar devices all over the place
quot; in the router/firewall. I usually set my devices to just
> 2 minutes and it works almost all the time. Most Cisco devices have a very
> long timeout of 3600 seconds.
>
> Leandro
>
>
> 2014-02-06 17:18 GMT+01:00 Mike Diehl :
>
>> Hi all,
>>
>> I have an
Hi all,
I have an SPA112 that in sitting behind a Ubee cable modem. The internet
link is solid, but the device becomes unreachable within a day or so of
being rebooted. Then the customer goes to reboot the device, they report
that all 4 lights are lit. The ISP reports that the device does respo
tration as well as database voicemail storage
(mysql). The database is on the same machine as the asterisk server.
Have we grown beyond the ability to host both the db and * on the same
hardware? Or is this a known issue wi
Does anyone know if Grnvoip is still in business, or what's going on with
them? I had an account with them, but they no longer terminate calls.
Mike.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
Rojas wrote:
> Hi
>
> You can do this,
> http://mike.eire.ca/2012/02/03/asterisk-1-8-vm-multiple-emails/
>
> If you are using asterisk 1.8
>
>
> On Wed, Sep 11, 2013 at 1:55 PM, Mike Diehl wrote:
>
>> Hi all,
>>
>> I've got a user who wants to
Hi all,
I've got a user who wants to receive voicemail notifications at two
different email addresses. I could probably setup an alias in
/etc/aliases, but then I'd have to manage that across multiple servers,
which I don't want to do.
Is there a way I can tell Asterisk to send to multiple addre
issue.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
> Sent: Wednesday, August 21, 2013 4:15 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-use
Hi all,
I've got a user with a couple of Cisco SPA303's. When I dial their phones
with a dial string like:
dial(sip/phone-a,300,rwkxttT)
The phone rings, as expected.
However after exactly 60 seconds, I get:
[Aug 21 02:09:56] -- Got SIP response 480 "Temporarily not available"
back from a
Fantastic! Thank you!
Mike.
On Thu, Aug 15, 2013 at 3:21 PM, Shane Young wrote:
> Quoting Mike Diehl :
>
>
>>
>> Is there a list somewhere?
>
>
> There is a list by state here:
> http://ww
Hi all,
I have a customer that tried to use the Texas One-Call number (a
toll-free call) to have the utility company come out and mark buried
pipes and cables. That call resulted in a recording telling her to
dial 811, instead.
So, as a service provider, how do I terminate a call to 811? In NM,
days ago without any incedent.
Now suddenly, the whole thing comes crashing down.
I also notice that while the backup is running, my other queries
block, which is probably why my peers disappear.
I'll be posting to the mysql list as well, but has any Asterisk user
seen this before?
TIA,
Yes, it should have, but I don't think it's turned on by default...
Anyway, this lower MTU setting is just temporary, so I'll just live
with it for now.
Thanks for your assistance.
Mike.
On Wed, Aug 7, 2013 at 2:28 AM, Steven Howes wrote:
> On 6 Aug 2013, at 19:28, Mike Dieh
We got it fixed! Our co-lo is in the process of doing a network
reconfiguration/relocation and had changed their MTU to 1400 during
the transition. Once we did the same, everything started to work.
Thank you all for your time and quick responses.
Mike.
On Tue, Aug 6, 2013 at 10:44 AM, Tim Nels
.conf. In fact, this server has
a virtually identical configuration to one that is already running.
(I sync the configurations using unison.)
I don't THINK this is a configuration issue. Any ideas, though?
Mike.
On Tue, Aug 6, 2013 at 10:08 AM, Warren Selby wrote:
> On Tue, Aug 6,
than that, there have been no configuration changes since the last
time this worked.
Any other ideas?
Mike
On Tue, Aug 6, 2013 at 4:36 AM, Jeremy Kister
wrote:
> On 8/6/13 5:30 AM, Mike Diehl wrote:
>>
>> sip show peer voice12 load
>>
>> This command just returns, wi
---
Name: db
DSN:db
Last connection attempt: 1969-12-31 17:00:00
Pooled: No
Connected: In use
I'm using 10.2.1. Also, I've noticed that command completion
doesn't work on the Asteris
all in the cdr table using
Uniqueid = 'server-1374906101.132305', my query doesn't find anything.
If I then ask the cdr database for the Uniqueid of the call in
question, I get something slightly different, such as:
server-1374906100.132304
I
this be disabled from the provisioning
file? Is there anything else I can do to prevent this?
TIA,
Mike Diehl.
--
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Thank you! That was very helpful.
Mike.
On Wed, Jul 10, 2013 at 7:38 PM, Matthew Jordan wrote:
>
> On Wed, Jul 10, 2013 at 5:35 PM, Mike Diehl wrote:
>>
>> Hi all,
>>
>> I'm contemplating an upgrade from 10.2.4 to 11.4.x. However, the
>> 1.8.x t
e less granular than the
release notes from 10.2.x to 11.4.x. I don't mind reading, but that
is almost as long as War and Peace!
Does such a document exist, or do I need to start reading..
TIA,
Mike Diehl.
--
_
-- Band
Thank you!
Mike.
On Tue, Jul 2, 2013 at 1:37 PM, Administrator TOOTAI wrote:
> Le 02/07/2013 21:06, Mike Diehl a écrit :
>>
>> [...]
>>
>> I was thinking that a TA with an FXO port might do the trick. But, I'm
>> not sure how to get the device to redirect
to get the device to redirect an incoming call on the FXO
port to a sip destination. Is this something that gets done in the
device's dialplan?
Does anyone have any insight into how to do this?
TIA,
Mike Diehl
--
_
-- Bandwidth a
Funny you should ask! I have an MP-202 in front of me right now that I'm
working on. When I get it working, I'll let you know. In the mean time,
what symptoms are you getting?
Mike Diehl.
On Mon, Jul 1, 2013 at 4:07 PM, David Wessell wrote:
> Does anyone have experience
On Thu, Jun 20, 2013 at 2:05 PM, Joshua Colp wrote:
> Mike Diehl wrote:
>
>> Hi all,
>>
>> I'm getting ready to setup SIP/TLS and SRTP. But I have a few
>> questions. The first one is that I was reading an article at:
>>
>> https://supportforum
ll for now.
Thanks in advance,
Mike Diehl.
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New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hell
nsfer to 4-digit extensions. Is there something I need to do for
the 450 to make this work?
Thank you in advance.
Mike Diehl.
--
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New to Asterisk? Join us
t of this dial is BUSY/21, which I understand. But, my perl
script isn't getting this value. I do the assignment to/from $result and
$cause because I use those values later in the script, but have to pass
them back to a global routine, as well.
Can anyone see what I'm doing wrong?
Mike Die
I went and checked; my database has the recording field defined as a
longblob.
Any other ideas would be most appreciated.
Mike Diehl.
"Danny Nicholas" wrote:
> IIRC blobs are normally set to a limit of 65 Kb. You may need to redefine
> as medium blob (16Mb) or long blob (4
with attachment intact.
My server is pretty busy, so I'm not really able to get a console output,
untill much later tonight, perhaps.
I'm thinking it's a tuning parameter on the mysql database that sets the
size of a blob, but I don't know.
Any help would be appreciated.
Mike Dieh
lover occurs, the phone's IP address will obviously
change. So, how can/should I configure this to minimize my customer's
down-time?
TIA,
Mike Diehl.
--
_
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On Thursday 03 May 2012 1:47:09 pm Paul Belanger wrote:
> On 12-05-03 01:45 PM, Mike Diehl wrote:
> > Hi all.
> >
> > I've got a perl script that connects to Asterisk's management interface
> > using Asterisk::AMI. So far, its proven to be very useful.
>
it 1;
=
When this scrip disconnects, the asterisk console simply says that the manager
logged off from 127.0.0.1 and got logged back on. No reall
error indication given.
Any su
just turn udptl off. I could expand the port
range, but I suspect that will just mask the situation.
What can I do to prevent this from happening?
TIA,
--
Take care and have fun,
Mike Diehl.
--
_
-- Bandwidth and Colocatio
On Tuesday 13 March 2012 4:04:31 pm Kevin P. Fleming wrote:
> On 03/13/2012 04:56 PM, Mike Diehl wrote:
> > On Tuesday 13 March 2012 3:45:18 pm Kevin P. Fleming wrote:
> >> On 03/13/2012 04:18 PM, Mike Diehl wrote:
> >>> I've set faxdetect to '
On Tuesday 13 March 2012 3:45:18 pm Kevin P. Fleming wrote:
> On 03/13/2012 04:18 PM, Mike Diehl wrote:
> > I've set faxdetect to 'yes' for the devices that I expect to be receiving
> > fax calls.
>
> 'faxdetect' is not a chan_sip configuration opt
up :
Mailbox : 7001@context
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit : 0
Dynamic : Yes
Callerid : "Mike Diehl" <5051234567>
MaxCallBR: 384 kbps
Expire : 172
Insecure : no
Nat : Always
ACL : Ye
n on the console that Asterisk heard a
fax. My extension simply rings and I answer it.
What am missing?
TIA,
Mike Diehl.
On Friday 24 February 2012 4:22:07 pm Kevin P. Fleming wrote:
> On 02/24/2012 05:20 PM, Mike Diehl wrote:
> > On Friday 24 February 2012 4:06:19 pm Kevin P. F
On Tuesday 06 March 2012 5:47:39 pm Patrick Lists wrote:
> On 07-03-12 01:28, Mike Diehl wrote:
> > I've been logging sip registrations from this IP address for 2 days now.
> > I've emailed the domain's admin, but nothing seems to come of it.
> >
> > I
route add -host 188.138.100.16 dev lo
Good bye. But it shouldn't come to this.
On Tuesday 06 March 2012 5:48:26 pm Matt Desbiens wrote:
> iptables -A INPUT --src 188.138.100.16 -j DROP
>
> On Mar 6, 2012 7:29 PM, "Mike Diehl" wrote:
> > I've been logging sip
I've been logging sip registrations from this IP address for 2 days now. I've
emailed the domain's admin, but nothing seems to come of it.
I've routed him into oblivion, but still, I think 50 requests a second for 2
days is a bit much.
Any ideas?
--
Take care and
On Friday 24 February 2012 4:22:07 pm Kevin P. Fleming wrote:
> On 02/24/2012 05:20 PM, Mike Diehl wrote:
> > On Friday 24 February 2012 4:06:19 pm Kevin P. Fleming wrote:
> >> On 02/24/2012 05:00 PM, Mike Diehl wrote:
> >>> On Friday 24 February 2012 3:17:04 pm Mike
On Friday 24 February 2012 4:06:19 pm Kevin P. Fleming wrote:
> On 02/24/2012 05:00 PM, Mike Diehl wrote:
> > On Friday 24 February 2012 3:17:04 pm Mike Diehl wrote:
> >> On Friday 24 February 2012 2:39:48 pm Kevin P. Fleming wrote:
> >>> On 02/24/2012 03:32 PM, M
On Friday 24 February 2012 3:17:04 pm Mike Diehl wrote:
> On Friday 24 February 2012 2:39:48 pm Kevin P. Fleming wrote:
> > On 02/24/2012 03:32 PM, Mike Diehl wrote:
> > > Hi all,
> > >
> > > I've got a user that has one phone number an wants to be a
On Friday 24 February 2012 2:39:48 pm Kevin P. Fleming wrote:
> On 02/24/2012 03:32 PM, Mike Diehl wrote:
> > Hi all,
> >
> > I've got a user that has one phone number an wants to be able to us it
> > for both voice and fax.
> >
> > When a fax call
--
Take care and have fun,
Mike Diehl.
--
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Hi all,
I'd like to know how I can turn off the "splash ring" voicemail waiting
indication on a PAP2T from the provisioning XML file. I can do it from the web
interface, but I need to do it on "a lot" of machines
TIA,
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Take care
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