Here's my attempt to explain a quick way of doing an auto dialer with
Scala and the Asterisk-Java library:
http://blogs.reucon.com/asterisk-java/2008/08/10/outbound_message_delive
ry_using_agi_and_ami_in_scala.html
Cheers,
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economi
org/development/mail-lists.html.
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> [EMAIL P
Hi Stefan,
I'd expect there's a Manager event that is fired when an IAX client
login happens. You could watch for that and initiate your call if
there's voicemail at that time.
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University
That's a good point. I don't know, honestly, if you can react to a SIP
register or an IAX login from within the dialplan. To anyone else:
Is there a way to act in the dialplan on a manager event?
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Busines
o.org/tiki-index.php?page=check_asterisk.
Cheers all,
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Beh
an be offered!
--
Jason Martin
Metrix Matrix, Inc.
785 Elmgrove Road, Building 1, Rochester, NY 14624
Office: 888-865-0065 Ext. 202
Mobile: (585) 705-1400
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write = system,call,log,verbose,command,agent,user
Le lundi 22 septembre 2008 ? 09:46 -0400, Jason Martin a ?crit :
> Hello,
>
> I'm trying to do some simple tests with AJAM on asterisk 1.4.17 and I'm not
> having much success.
>
> Right now the http server just li
r tells me that it's "usually something else", and that the errormessage
is not that descriptive.
What can I do to get more/better debugging info? I can't figure out what's
wrong.
Thanks!
- Martin
( my iax.conf and extensions.conf on http://pastebin.com/mb0020
ugh.
I doubt it. It has been working fine for a while, and others report IAX2
working fine.
- Martin
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Register Now: http://w
Inter Asterisk eXchange (Ver 2) 0
1 modules loaded
> Did you build and load ztdummy (assuming you have no Zaptel/Dahdi cards?
No - but i don't use MeetMe?
Thanks,
Martin
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ponse upon a timeout with no digits pressed. I'd also
encourage you to check out the Asterisk-Java mailing list via
http://asterisk-java.org/development/mail-lists.html.
Cheers,
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
o your
> provider for instructions on how to setup the trunk.
That was indeed the problem. I added this to iax.conf:
[myprovider]
type=friend
username=88821268
secret=xxzzyy
host=s1.core.myprovid.er
And used this in extensions.conf:
exten => _ZXXX,2,Dial(IAX2/myprovider/${EXTEN:
so the solution
really has to work.
Does anyone else on the list have a PRI to VoIP failover setup that's
worked for them in a high volume environment?
Thanks!
Jason Martin
Metrix Matrix, Inc.
785 Elmgrove Rd, Bldg 1
Rochester, NY 14624
Office: 888-865-0065 x202
F response as my modem does not
seem to know the network names. I also needed to increase the storage size for
this field to prevent data corruption.
The problem is I can't dial out or accept incoming calls.
-- Executing [3...@home:2] Dial("SIP/martin-007ab0c8",
"sebi/h
I put in "mwisendtype=nofsk"
in chan_dahdi.conf anyway, and all features like faxdetect and
transfer are turned off.
Has anyone else experienced this issue and fixed it?
Thanks.
Jason Martin
Metrix Matrix, Inc.
785 Elmgrove Rd, Bldg 1
Roches
fine with previous versions of asterisk.
Jason Martin
Metrix Matrix, Inc.
785 Elmgrove Rd, Bldg 1
Rochester, NY 14624
Office: 888-865-0065 x202
Mobile: 585-705-1400
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is?
>
If you want to send me your patch direct I will make it available through my
website http://www.mycrofters.com and we could also use the forums there to
continue the discussion.
Martin
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Fax receive not successful - result (13)
Unexpected message received.
For detailed logs please take a look of http://www.pastebin.ca/1634790
Cordialmente,
Martin Cabrera
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asteris
Hello again Asterisk people.
I am running Asterisk 1.42 on an old PowerPC ibook. I have had this
deployed for several years now, with pretty good results.
Recently I added a callerid service to my landline (qwest).
I am using the audiocodes MP114 2fxo/2fxs gateway, which is an
outstanding
On Nov 6, 2009, at 5:14 AM, John A. Sullivan III wrote:
> On Thu, 2009-11-05 at 20:57 -0800, Martin Joseph wrote:
>> Hello again Asterisk people.
>>
>> I am running Asterisk 1.42 on an old PowerPC ibook. I have had this
>> deployed for several years now, with pretty g
Ok I am replying to myself, because I still don't have this figured
out,, but I think I have more info.
On Nov 5, 2009, at 8:57 PM, Martin Joseph wrote:
>
> Hello again Asterisk people.
>
> I am running Asterisk 1.42 on an old PowerPC ibook. I have had this
> deployed f
sip.conf so my
incoming call handling peer is at the very end.
Pretty wacky.
I am hopefully back on the road though with working caller ID as well.
Marty
On Nov 14, 2009, at 11:10 AM, Martin Joseph wrote:
> Ok I am replying to myself, because I still don't have this figured
>
ly using. So my question should I use hpec
or oslec with my TDM400 card? I also tried to recompile dahdi to use
oslec (before I found that Digium had hpec) but then I get an error
message that the source of my kernel cannot be found so I can never
actually compile a new version of dahdi.
AsteriskNow use CentOS 5 and it comes preinstalled with dahdi and
asterisk with the freepbx GUI interface and it seems to be missing all
the dev packages
Martin
On 2009-11-17, at 02:19, Olivier wrote:
2009/11/17 Martin Roy
I was previously using an old computer running Asterisk 1.2
On 2007-02-22 04:22:20 -0800, "Frederico Madeira" <[EMAIL PROTECTED]> said:
Hi guys,
My asterisk is show me some errors on line registration.
This message appear on console: Request to schedule in the past?!?!
What it mean ?
Thanks.
I see this message all the time on my lowely powerPC mac (
On 2007-03-24 01:53:16 -0700, Edoardo Serra
<[EMAIL PROTECTED]> said:
Hi Francois,
[EMAIL PROTECTED] ha scritto:
Hi men,
I have already encountered some issue like this with few switches (very
known great brand) which doesn't like VoIP traffic !
I also have switches of a very known gre
On 2007-03-23 14:37:18 -0700, "Tom Lynn" <[EMAIL PROTECTED]> said:
Now I know where they've been spending my remaining balance...
I still use Sellvoip as my primary terminator, and have found the call
quality to be superior to any other ITSP from my location (Seattle).
I agree completely
I am having problems with my zaptel channels on my fresh install of Asterisk
1.4.2 on Fedora core 6.
I have a new Digium TDM400P with 2 FXO modules.
Both dmesg and ztcfg -vvv show the FXO modules loading correctly:
-
Zaptel Version: 1.4.1
Echo Canceller: MG2
Configuration
Just a warning for you all that are using Nokia series E phones for SIP
function.
I updated my phones firmware today using the Nokia Updater, and now
the SIP functionality, which previously worked pretty well is
completely broken.
The phone no longer registers with asterisk, although it dis
On 2007-04-17 00:53:56 -0700, Dinesh Nair <[EMAIL PROTECTED]> said:
On Mon, 16 Apr 2007 20:14:40 -0700, Martin Joseph wrote:
The phone no longer registers with asterisk, although it displays the
little icon as though it has, and it doesn't even seem to try to pass
calls to aster
On 2007-03-26 01:46:40 -0700, "Salvatore Giudice"
<[EMAIL PROTECTED]> said:
This is a multi-part message in MIME format.
I opened up a ticket with them, but I'm not holding my breath. I think it's
time to start moving my DID's before the inbound stops working.
That seems like it was probab
machine and our
primary, active one. We can't really give up the PRIs without some
downtime, so we're specifically interested in solutions that allow a
primary machine to remain in operation while testing a secondary, and
without using up the PRI circuits for testing (but we want to test ou
Hello again gurus.
I have been using Asterisk with great results going on a couple of
years now.
My primary box is running asterisk 1.42 built from a tar ball on Mac
OSX 10.4.9.
I have a very odd issue that I cannot seem to nail down, which is
related to my Nokia E60 SIP phone.
I use
6:MMI-Y:200705081051010077',
'uniqueid' '51010077',
'userfield' '',
'MMI_field' 'not found'
Issue #2: When a call is not answered, a record of that call is written to the
database, but uniqueid is left blank. The next time a call isn
On May 14, 2007, at 12:34 PM, Tim Panton wrote:
On 14 May 2007, at 17:50, Martin Joseph wrote:
Hello again gurus.
I have been using Asterisk with great results going on a couple of
years now.
My primary box is running asterisk 1.42 built from a tar ball on
Mac OSX 10.4.9.
I have a
If you use edit the config files on a trixbox system like you would on
an * box, any time you reboot or hit the red update bar, it will reset
the files to what the gui has. The only files you can edit on a trixbox
system are the _custom.conf files. This may be the issue with the time out
Martin D
the word proxy!). Figured I'd send this out in case
someone hadn't seen it.
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
> -Original Message-
> From: [EMAIL PROTECTED]
> [mai
to have but I never had to deal
with patch before. I usually just take the release version of asterisk and
install it as is.
P.S. I would like to keep the version 1.4.21 because it's the last version that
I know of that use Zaptel by default instead
/DAHDI chunk size and that directly affects system load.
Second question - the document explains how to change the chunk size for
Sangoma hardware. Is there a general way to do that for DAHDI?
Thanks is advance!
Jason Martin
Metrix Matrix, Inc.
785 Elmgrove Rd, Bldg 1
Rochester, NY 14624
Office
s anyone got any hints from installing trader turrets (for
instance) about what dahdi config I need for this dedicated type of
T1?
Thanks
Martin
:-)
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Dear Colleagues,
I installed a Wellgate 3804A and overnight lines on all this with frying,
putting other lines Wellgate 3804A is well, so I guess it's a problem the first
team which is already out of warranty, anyone know how can I fix this? or where
to send it in or capital Buenos Aires to fix
Hi,
checkout new open source voipmonitor.org SIP packet sniffer. I've
developed it for my telco company and I've decided to share it.
Testing and contributions are welcome!
VoIPmonitor is open source live network packet sniffer which analyze
SIP and RTP protocol. It can run as daemon or analyzes a
Hello,
I've choosen only MOS-LQE because it is calculated only on network
parameters, which is loss, burstinnes and delay (which is converted to
loss by jitterbuffer simulator). It does not takes into account voice
(payload). There is no effective objective methods (today) which
predicts MOS. Only
On Sat, May 8, 2010 at 2:34 PM, mosbah abdelkader
wrote:
> Thank you Martin,
>
> So the MOS-LQE does not inform bout payload itself but predicts the MOS
> based on networks metrics
yes exactly. LQE is Listen Quality Emodel (E-model is parametric model
which takes into accou
On 8.5.2010 00:40, Jeff Brower wrote:
> Martin-
>
>
>> checkout new open source voipmonitor.org SIP packet sniffer. I've
>> developed it for my telco company and I've decided to share it.
>> Testing and contributions are welcome!
>>
>> VoIPm
idea what is going on, any parameter in the zap config files that
would fix that?
Thanks
/Martin
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?
Michael Martin
Systems
Engineer
Netranom Communications
*
email: [EMAIL PROTECTED]
(
office: 304.562.4700
h
mobile:
304.419.1510
:
web: www.netranom.com
.
Best regards,
Martin
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.
Anyone found a way to make it work?
Thanks
Martin Roy
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ossible. So I'm wondering if there's any one out there that found
a phone that can be change to french.
Thanks
Martin Roy
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hanks
Martin Roy
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Do you have access to a T-1 analyzer? You more than likely have a 'dirty'
T-1 line that is out of spec based on the length of the run.
Nik
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bonzo Armstrong
Sent: Monday, June 21, 2004 5:43 AM
To: [EMAIL PRO
On Mon, Jun 21, 2004 at 12:17:38PM -0500, Nik Martin wrote:
>
> Do you have access to a T-1 analyzer? You more than likely have a
> 'dirty' T-1 line that is out of spec based on the length of the run.
Sadly, none that I'm aware of, but I'll ask around. I could
Carlos Medina wrote:
> Hi, i have a call center which receives many calls at day. Those
> calls are stored in a directory in my asterisk server as WAV files.
> The problem is that each call is divided in 2 files: an IN.WAV file
> and OUT.WAV file. The OUT.WAV file is what im speaking to other
> per
GIBERT Frédéric wrote:
> Hello Adam,
>
> I'm interested by this solution, but can you please give me more info
> because I don't know how to generate calls with asterisk and the
> spool directory. How don't know wich files do I need to use.
>
> Thanks.
> Fred
Look in your ./asterisk directory,
Michael Welter wrote:
> Jason A. Pattie wrote:
>> Robert Hajime Lanning wrote:
>>
>>> Echo echo ech ech ec ec e e . .
>>>
>>> :)
>>>
>>>
>>>
What's the importance of the impedance matching in a FXO interface
?
>>
>>
>>
> My experience is with excessive buzz and hum on the line. W
Chris Shaw wrote:
> Ok I have googled and googled and combed through the wiki for an
> answer to this and have come up empty. What I'm finding is that when
> a user changes their VM password, it is saved somewhere like maybe
> the CSV database or something because when you log in, the new
> passwor
Bonzo Armstrong wrote:
On Wed, Jun 23, 2004 at 08:24:03PM -0700, Bonzo Armstrong wrote:
On Mon, Jun 21, 2004 at 03:04:52PM -0500, Nik Martin wrote:
Try this if possible. Connect the channel bank to * via the 400' cable, but
in the same room as the * box, with all the cable coiled on the
Rich Adamson wrote:
>
> From: Nik Martin <[EMAIL PROTECTED]>
> Subject: RE: [Asterisk-Users] FXO impedance matching
> Date: Wed, 23 Jun 2004 11:02:00 -0500
> To: [EMAIL PROTECTED]
>
>
>> Michael Welter wrote:
>>> Jason A
Jim Gottlieb wrote:
> Hi all. I've been trying to build some new systems, and no matter
> what I do, if I load the zaptel and tor2 drivers, the system panics
> within an hour, even with no traffic.
>
>
> A typical Call Trace from the panic message looks like:
>
> wait_on_irq, [kernel] 0xde
>
You replied to a message with the subject of:
Re: Do people actually answer questions here?
And then changed the subject and started typing. This has wreaked havoc on
everybody's threaded readers, and made your question impossible to reply to.
You need to start a new message in your mail app and s
t;H323:10", "IAX2/[EMAIL PROTECTED]/s|40|r")
in new stack
The number is missing in the H323 and replaced
with an s instead. Anyone who knows why this is
happening?
Best regards
Martin Kiefer
dge betveen the AIX and the H323.
I don't see how the GnuGK can translate the H323 call to either a SIP or
AIX call?
Best regards
Martin Kiefer
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> So then don't tell the EP to register with a GK, just send
> calls to an
> H.323 Gateway, Asterisk.
>
And that is exactly what I am trying to do. But with no luck.
Best regards
Martin Kiefer
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all T1/E1 boards do
regards
Martin
On Wed, 30 Jun 2004, Gonzalo Mateos wrote:
> Hi there,
>
> I'm quite new to asterisk and digium hardware. I needed to know which of the digium
> cards supports E&M signaling?.
>
> thnaks,
> Gonzalo
>
>
> ---
> Outgoi
Hi there
I am trying to do something similar to this. My setup looks like this:
Cisco 7970G IP phone > Cisco Call Manager 3.3.(4) using H323 > Asterisk
IAX > Provider supporting IAX
So a sample configuration on how to translate H323 to IAX is very
welcome here too.
Best regards
Mart
{TRUNKMSD}})
I am using the default settings in my oh323.conf. Am I missing something
in this file?
Best regards
Martin Kiefer
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Weee, thank you so much for that help.
I can now make calls from my Cisco Call Manager to Musimi :-)
/Martin
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Soren Rathje
> Sent: Saturday, July 03, 2004 3:33 AM
> To: [EMAIL PR
We're using the Quad-BRI card from Junghanns.NET with corresponding
drivers (bristuff 0.0.2).
The driver tries to patch asterisk libpri, which fails for current
version.
Anyone got an idea what'S the latest version of asterisk / libtri usable
with the Quad-BRI Card?
Than
work?
Same reaction using the innovaphone ip400 gatekeeper and using gnugk.
Asterisk version is 0.7.2 release.
Thanks, Martin
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't compile against
the versions usd by bristuff 0.2.2.
Is it possible to combine older libtri with cvs-head asterisk or is that
just asking for trouble?
Thanks, Martin
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> How about running a current (cvs -head) version of Asterisk?
Would love to and of course tried to: no go because of Junghans Quad-BRI
ISDN Card, no driver for cvs -head.
Bye, Martin
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h
> If you can't wait you can use the patch from someone who merged the
> bristuff patch with a more recent version of cvs head...
>
> This one:
> http://capi4linux.thepenguin.de/download/asterisk/bri-stuff-0.
> 0.2a-pp.tar.gz
Thanks for that pointer, I'll
Hi Torsten,
> I think I have the same problem as Martin Bene mentioned in
> http://lists.digium.com/pipermail/asterisk-users/2004-January/
> 034521.html
> Since I found no further information about this I'd like to
> ask wether you know what the reason for this problem is an
Bisker, Scott (7805) wrote:
>
> Depending on your familiarity with linux, the learning curve could be
> steep and prove frustrating considering everything else you'll be
> dealing with (new network infrastructure, new computers, new servers,
> new telco/data circuits). Less expensive components d
work properly.
regards
Martin
On Wed, 7 Jul 2004, Gelson Dias Santos wrote:
> Steven Critchfield wrote:
>
> > On Tue, 2004-07-06 at 17:52, Ruben Fagundo wrote:
> >
> >>I have an easy question. I setup Asterisk with a TDM400 w/ 4FXO ports
> >>and I have the follo
Philipp von Klitzing wrote:
> Hi!
>
>> Does anyone have a current, stripped linux distro which has only
>> asterisk and net drivers?
>
> Look here: http://www.voip-info.org/wiki-Asterisk+installation+tips
>
> and you'll find a link to the "Asterisk Live! CD-ROM".
>
> If you have a moment I gues
Ing. Angel Gomez wrote:
> Thank's to all.
>
> - The card came WITHOUT ANY documentation, it was not buy directly
> from digium, they did not have any in stock.
> - I usually go thru all the messages of this user list, maybe I
> overlook at one with the same question.
> - Th
Eugen Cristea wrote:
Hi,
I would like to set two separate asterisks to talk to
each other.
Any suggestions?
I'm a "baby" asterisk fan, only started to play two
weeks ago, first managed to use kphone with asterisk
and a X100P card that is up and running as well.
Thanks,
Eugen
prepare to get flamed
ving phone.
According to RFC 3261 SIP uses UTF-8 encoding. Shouldn't asterisk
convert these characters from ISO-8859-1 to UTF-8 before passing
them to SIP devices?
Best regards
martin
--
Martin A. Blatter | lic. oec. publ. Wirtschaftsinformatiker | IT-Leiter
OLMeRO AG | Europastrasse 30 | CH-8152 Gla
CHS wrote:
> is there a well-written, easy to follow, voicemail setup guide for
> asterisk?
>
> for now I don't care about understanding HOW voicemail works, I would
> just like to see a quick-start guide, similar to the VOIP quickstart
> guide on the voip-info.org website.
>
> I'm about to go
Hello again,
sorry for the delay in replying; I've been off for some weeks at a
customer's offices and couldn't read my email at work...
ePyron Felix Deierlein wrote:
Hello Martin,
how would you like to integrate? PRI (E1) or BRI (ISDN)?
Besides of making calls with
Fletcher Bonds wrote:
> Hello all
>
> As of 5pm PST today (7/13), I pulled Asterisk down off of
> cvs.digium.com:/usr/cvsroot and tried to compile it on Linux ES 2.1
>
> Actually, I pulled down zaptel, libri & asterisk and compiled them in
> that order as per my install guide.
>
> When I try t
Gabriel Millerd wrote:
>>> Is there a magic 'fan card' that has a power out that people are
>>> using?
>>
>> This may work for you.
>>
>> http://www.thermaltake.com/products/subzero/subzero4g.htm
>>
> you lost me, its a processor cooling device. it doesnt provide
> any power that could be us
Alessio Focardi wrote:
> Hi,
>
> it may be off topic, in case excuse me.
>
> I need to get a USA phone number, possibly a Florida one.
>
> I would like to use my actual sip phone to connect, also I would love
> montly flat rates on calls.
>
> Have you got a provider to suggest that will allow
Fletcher Bonds wrote:
>>> Can anyone tell from that error if I'm missing something or what
>>> the problem may be?
>>>
>>> Thanks a bunch
>
>> Yep, you need bison
>
> I have bison.
> # bison -V
> GNU Bison version 1.28
>
> Is it expecting a different version than that?
Are these phones any closer to being avaliable?
- Original Message -
From: "Adam Hart" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, April 01, 2004 5:38 PM
Subject: Re: [Asterisk-Users] Virbiage Phones - Vapourware??
> Aaron Martin wrote:
&g
Hi all,
I noticed that all incoming calls come from the user "[EMAIL PROTECTED]", so
I just can't hit the "Call" button on my SJphone for Linux to return the
call...
Is there any way to configure Asterisk to show the real [EMAIL PROTECTED] ?
T
.
Thanks
Martin
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as the "Name" part of the caller ID. How can I get and set this
information within Asterisk?
Thanks for any pointers.
regards
martin
< Message type: INFORMATION (123)
< [< [28< [28 08< [28 08 46< [28 08 46 52< [28 08 46 52 2e< [28 08 46 52 2e 20< [28 08 46 52 2e
CDR table in a database called Asterisk. Conf files etc are set. I even
recompiled Asterisk. Any pointers would be greatly appreciated.
Martin
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To
s still don't seem to show up
in asterisk (they *do* show up in the pri debug output). Any
ideas why?
Best regards
martin
Klaus-Peter Junghanns wrote:
Hi,
you can take a look at how bristuff does this (it only has to be enabled
in chan_zap to actually forward the display IE, uncomment line 8
I installed Postresql and then recompiled Asterisk. I understood that
Asterisk would see Postresql on the recompile and add it. Is there a way of
checking?
Martin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen R.
Darragh
Sent: Tuesday, July 27
Title: Message
Greetings
Does
anyone have a ADSI config file for an Astra (Sayson) 480e phone. I am using
the sample asterisk.adsi file but if anyone already has a modified working file
that they would like to share, could you let me know.
Thanks
Martin
Thanks for your help
The cdr_pgsql.so was not there. Do I change the mods line to
MODS=cdr_csv.so; cdr_pgsql.so
Ie. Do I add a semi-colon or not.
Thanks
Martin
Hi -
Check /usr/lib/modules to see if cdr_pgsql.so is in there.
If not, edit the Makefile in the asterisk/cdr directory and add
4\x30"
Then recompile and press the Vmail button on your phone. It should
automatically download the script and then you have a bunch of new buttons
to play with!
On a side note, I am tring to enhance the ADSI programing in the orignal
script. Did your supplier give you any help with addition
Just a died question. Will all of the sessions be recorded and made
available?
Martin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E.
Johansson
Sent: Thursday, July 29, 2004 10:34 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk
I am having a problem with getting voice mails, even when the caller hangs
up before getting to the recording prompt. If I call my number, even if I
hang up the second I get the "I'm not in" recording, it still generates a
voicemail. Is there a way aroun
frustrating is that
a caller who hanges up during a "please leave message" prompt, still
generates a message, even if they hang up without leaving a message. Is
there a way improving this?
Thanks for your help.
Martin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PRO
aven't got
it to register properly with Asterisk. Released a little to early I think.
However, it is a very professional looking phone and once the bugs are out
of it, it seems very promising at a good price.
Martin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTE
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