- Original Message -
From: Robert Rozman [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, March 20, 2008 7:40 PM
Subject: [asterisk-users] 423 Interval Too Brief and expiry settings
insip.conf
Hi,
I'm getting this error when registering with SIP server using
Hi,
I have a problem with DIAL.
The scenario is this:
1. Asterisk will dial a number in a call list
2. called party picks up the call and hears a prompt asking if they want to
pick up ( this is done through M(marco) option in DIAL)
3. if called party does not want to pick up, go to the next
2008/4/1, Jean-Denis Girard [EMAIL PROTECTED]:
Hi,
Olivier a écrit :
Hi,
Is it possible to both use Digium B410P and bristuff'ed 1.4 Asterisk,
now ?
I've heard BRI support in Asterisk is about to change with 1.6 but I'm
not sure I understood what the plan is.
If someone has a
Hi,
has anyone connected Asterisk to Cirpack softswitch sucessfully ? Any howto
or more info about needed Asterisk SW and setup ?
Thanks in advance,
regards,
Rob.
___
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On Wed, Apr 2, 2008 at 9:11 AM, Robert Rozman [EMAIL PROTECTED] wrote:
Hi,
has anyone connected Asterisk to Cirpack softswitch sucessfully ? Any howto
or more info about needed Asterisk SW and setup ?
Thanks in advance,
regards,
Rob.
We have a trunk supplier that uses a Cirpack
On 10:11, Wed 02 Apr 08, Robert Rozman wrote:
Hi,
has anyone connected Asterisk to Cirpack softswitch sucessfully ? Any howto
or more info about needed Asterisk SW and setup ?
Yes, it works fine.
Where do you get stuck ?
It's basically a normal sip connection setup.
--
Michiel van Baak
Hi,
I just upgraded from 1.2 to 1.4.
In 1.2, when I did a show uptime I used to see a
second line telling me the time since the last reload.
Has this been removed in 1.4?
The following is the output of my two test boxes:
Connected to Asterisk 1.4.18.1 currently running on
voip2 (pid = 10605)
In article [EMAIL PROTECTED],
Pete Kay [EMAIL PROTECTED] wrote:
I have a problem with DIAL.
The scenario is this:
1. Asterisk will dial a number in a call list
2. called party picks up the call and hears a prompt asking if they want to
pick up ( this is done through M(marco) option in DIAL)
On 01:40, Wed 02 Apr 08, Vieri wrote:
Hi,
I just upgraded from 1.2 to 1.4.
In 1.2, when I did a show uptime I used to see a
second line telling me the time since the last reload.
Has this been removed in 1.4?
The following is the output of my two test boxes:
Connected to Asterisk
Peder @ NetworkOblivion wrote:
That makes sense. A call from 729 to 711 would require one encoder and
one decoder, right?
So if you have 10 licenses, is it 10 total encoders+decoders, or 10
calls (some may require encode, or decode, or both)? Because I had 10
licenses, but my
You can get much better results (close to 56k reliable connections
sometimes) by using a Xorcom FXO Channelbank - You need recent enough
drivers so that the Xorcom internal clock can be synced to Zaptel;
This removes/reduces jitter and frame slippage, and allows a modem to
operate much more
Dear Tony,
Thank you very much for your suggestion.
The thing is that ${DIALSTATUS} returns ANSWER regardless of whether the
called party hangs up or not.
For instance, when called party is being asked by Asterisk whether he/she
wants to pick the call, ${DIALSTATUS} returns ANSWER.
In the case
Greetings list,
Not exclusively asterisk-related, but I've noticed the CentPBX site has been
offline the last few days. Anyone know the reasoning behind that, and more
importantly, is anyone mirroring it?
Thanks in advance.
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
For
Mojo with Horan Company, LLC wrote:
P.S. If you can't dial seven digit numbers in your area, but you miss
it, you can restore that behavior if you feel like selecting a default
area code:
exten = _NXX,1,Dial(Zap/1/907${EXTEN},,TWK)
Here, if I dial a seven digit number, asterisk
--- Michiel van Baak [EMAIL PROTECTED] wrote:
On 01:40, Wed 02 Apr 08, Vieri wrote:
Hi,
I just upgraded from 1.2 to 1.4.
In 1.2, when I did a show uptime I used to see a
second line telling me the time since the last
reload.
Has this been removed in 1.4?
The following is
Sorry for top-posting, but seems everyone on this thread did so.
Also that would be my suggestion for now - call queue with periodic-announce.
However i see that this would make nice architectural improvement -
allow inject sound files into MoH stream. This would be useful for
example in call
[EMAIL PROTECTED]:4] AGI(SIP/202-b654e668,
recordingcheck|20080402-143454|1207139694.24) in new stack
-- Launched AGI Script /usr/share/asterisk/agi-bin/recordingcheck
recordingcheck|20080402-143454|1207139694.24: Outbound recording not
enabled
-- AGI Script recordingcheck completed
I would like to know if the following misdn warnings
are relevant.
Currently, I don't need echotraining.
However, I took a quick look at the * source code and
l1watcher_timeout seems to be defined (echotraining
was not found). Currently I'm setting
l1watcher_timeout to 0 which is default (so I
I've been a happy user of asterisk for over a year just for a small home
setup (a Digium TDM400P with one POTS line and three internal extensions
plus a couple of SIP phones). I recently moved from running Fedora Core
6 running * 1.4.1 compiled from source and zaptel 1.4.7 to Fedora 8,
using the
CentPBX has bit the dust I believe.
-D
From: [EMAIL PROTECTED] on behalf of Chris Bagnall
Sent: Wed 4/2/2008 7:12 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] CentPBX mirror?
Greetings list,
Not exclusively
Jerry Geis wrote:
On Tue, 2008-04-01 at 13:24 -0400, Jerry Geis wrote:
/ I call into the dialplan and try to play demo-congrats and I hear
nothing.
// // Firewall is disabled. // Everything is on the 192.168.1.X
network for this simple configuration.
// The tftp server is giving the
On Wed, 2008-04-02 at 07:28 -0600, Greg Woods wrote:
[Apr 2 07:13:48] WARNING[24249] chan_zap.c: Detected alarm on channel
4: No Alarm
[Apr 2 07:13:48] NOTICE[24242] chan_zap.c: Alarm cleared on channel 4
core set debug 3 doesn't give any more detail.
I have tried stopping asterisk,
On Wed, Apr 2, 2008 at 10:28 AM, Greg Woods [EMAIL PROTECTED] wrote:
I've been a happy user of asterisk for over a year just for a small home
setup (a Digium TDM400P with one POTS line and three internal extensions
plus a couple of SIP phones). I recently moved from running Fedora Core
6
Greg Woods wrote:
I've been a happy user of asterisk for over a year just for a small home
setup (a Digium TDM400P with one POTS line and three internal extensions
plus a couple of SIP phones). I recently moved from running Fedora Core
6 running * 1.4.1 compiled from source and zaptel 1.4.7 to
Hello,
We achieve this using an AGI script in the VICIDIAL project for our
version of inbound queues. You start MoH then when you stream a sound
to the channel it will stop MoH then after the sound is done you start
MoH back up again. Probably a bit more involved than what you want,
but it dose
In article [EMAIL PROTECTED],
Pete Kay [EMAIL PROTECTED] wrote:
Dear Tony,
Thank you very much for your suggestion.
The thing is that ${DIALSTATUS} returns ANSWER regardless of whether the
called party hangs up or not.
For instance, when called party is being asked by Asterisk whether
A w in the D() string will wait .5 second. Example:
Dial(Zap/g1/5551212,,D(ww668))
If you are dialing out of an FXO or FXS signaled port, you can add w
to the dial string to wait .5 second. Example: Dial(Zap/g1/ww5551212)
Pete Kay wrote:
Is there anyway to have Asterisk to wait for 1 second
That does indeed sound a bit odd. I've run 12-48 FXS ports from a single molex
connector with Sangoma hardware. Try testing your power supply with a
multimeter to ensure its putting out the proper voltage. I would not trust the
extnernal AC adapters as I've found they typically have voltage
Hi
I just want to know if anyone have problems with server DELL 1600,
Like: Hangup Call.
Thanks
Ruben
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
If you have subversion installed on your server, could you try using
this version of zaptel:
http://svn.digium.com/svn/zaptel/branches/[EMAIL PROTECTED]
Not Found
The requested URL /svn/zaptel/branches/[EMAIL PROTECTED] was not found on this
server.
Thanks Matthew!
Now I can start looking for a workaround ;)
/hanna
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Fredrickson
Sent: den 28 mars 2008 16:39
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
Thanks for your answer.
I've found out that the zaptel drivers don't support Call Deflection at the
moment and in Sweden the callerid can be set to anything different than the
phonenumber of the caller.
Have to find a workaround :)
/hanna
-Original Message-
From: [EMAIL PROTECTED]
Ciao FaberK,
Hi folks,
I'm trying to install asterisk with radius cdr support.
I got freeradius up and running, so following radius instructions
inside asterisk source package, I've installed radiusclient-ng and
relative headers.
But when I start configure(asterisk 1.4.18.1) I got:
Hi all, I seem to only be getting (1) call to sip_write() in
channels/chan_sip.c
I have a very simple setup. one server (no cards) 2 polycom IP 330 phones.
Server is 192.168.1.150 and phone is DHCP. Nothing else on the network.
No firewall is enabled.
I call into the dialplan with:
exten =
CentPBX has bit the dust I believe.
Thanks. Any suggestions for a suitable FreePBX-based alternative with kernel
support for a Dell R200 (it's usually the SAS controller that causes the
problem)? I've tried PBX-in-a-Flash without success, and Trixbox is rather too
customized for what I'm
Kevin P. Fleming wrote:
Mojo with Horan Company, LLC wrote:
P.S. If you can't dial seven digit numbers in your area, but you miss
it, you can restore that behavior if you feel like selecting a default
area code:
exten = _NXX,1,Dial(Zap/1/907${EXTEN},,TWK)
Here, if I dial a
Hi.
I am trying to install asterisk_srtp.
I started by installing zaptel-1.4.9.2 and then I run the configure of
asterisk_srtp. In the menuselect of asterisk, the chan_zap in Channel
Drivers is always unselected(XXX). The only dependency that I don't
seem to have is zaptel_vldtmf.
Where can
Olivier a écrit :
Would you mind if I asked you this :
- Which card did you include in your home system ? Are you using an ISDN
BRI access ?
This is a basic BRI card with HFC chipset (Bewan Gazel 128)
- Is libpri necessary for ISDN BRI access ? I thought libpri was mostly
dedicated to
FreeBSD 7 asterisk and asterisk-gui from ports should make it quite easy to
get going, less then an
hour to install base OS and build ports, I did an install for a client in
less then 2 hours
On Wed, Apr 2, 2008 at 11:10 PM, Chris Bagnall [EMAIL PROTECTED] wrote:
CentPBX has bit the dust I
just an afterthought there is askozia also... though also FreeBSD based with
Web GUI
On Wed, Apr 2, 2008 at 11:10 PM, Chris Bagnall [EMAIL PROTECTED] wrote:
CentPBX has bit the dust I believe.
Thanks. Any suggestions for a suitable FreePBX-based alternative with
kernel support for a Dell
Greg Woods wrote:
If you have subversion installed on your server, could you try using
this version of zaptel:
http://svn.digium.com/svn/zaptel/branches/[EMAIL PROTECTED]
What's the svn command to fetch it and I'll try it.
svn co http://svn.digium.com/svn/zaptel/branches/[EMAIL
On Apr 1, 2008, at 5:22 PM, [EMAIL PROTECTED]
wrote:
Can the volume of the recorded voice mail message be changed? If
so, what I am doing wrong? Any input would be greatly appreciated.
Thanks.
I had a similar problem in our setup where we e-mail the recorded
messages to e-mail
I have a some setup scripts that use centos 4 or 5 and freepbx you are
welcome to use them.
Jonn
http://www.taylortelephone.com/asterisk/
Chris Bagnall wrote:
CentPBX has bit the dust I believe.
Thanks. Any suggestions for a suitable FreePBX-based alternative with kernel
support
The Asterisk development team has released version 1.4.19 of Asterisk and
1.6.0-beta3 of Asterisk-addons.
The new Asterisk-addons release contains a few bug fixes over the previous
version.
http://svn.digium.com/view/asterisk-addons/tags/1.6.0-beta3/ChangeLog?view=markup
Asterisk 1.4.19
Thanks. I will give this a try.
-John
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Date: Wed, 2 Apr 2008 09:29:48 -0700
Subject: Re: [asterisk-users] Voicemail- Recorded Mesage Low Volume
On Apr 1, 2008, at 5:22 PM, [EMAIL PROTECTED]
wrote:
Can the volume of the
Shaun Ruffell wrote:
svn co http://svn.digium.com/svn/zaptel/branches/[EMAIL PROTECTED]
zaptel-1.4-4122
Thank you, I will try that tonight when I get home and report back.
--Greg
___
-- Bandwidth and Colocation Provided by
Jerry Geis wrote:
On Tue, 2008-04-01 at 13:24 -0400, Jerry Geis wrote:
/ I call into the dialplan and try to play demo-congrats and I hear
nothing.
// // Firewall is disabled. // Everything is on the 192.168.1.X
network for this simple configuration.
// The tftp server is giving the
Not sure why I missed this earlier in the year but has anyone had a look
at OpenFrame home handset?
Any comments?
http://www.pcmag.com/article2/0,2704,2246158,00.asp
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357
+61-2-9016-5642 (Sydney in-dial).
For interested parties
It would appear I didn't need to had anything between the
loadInformation/loadInformation tags in XMLDefault.cnf.xml and/or
SEP{MAC Addr}.cnf.xml to force the upgrade, although the correct version of
firmware needs to be present between the tags once it is converted.
On Tue, Apr 01, 2008 at 01:44:39PM -0400, Jerry Geis wrote:
I have no card in this unit at this time.
lsmod shows ztdummy loaded.
Just to make sure that this is not the problem, what's the output of:
zttest -c 3
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL
On Tue, Apr 01, 2008 at 01:44:39PM -0400, Jerry Geis wrote:
/ I have no card in this unit at this time.
// lsmod shows ztdummy loaded.
/
Just to make sure that this is not the problem, what's the output of:
zttest -c 3
--
When running this nothing comes back...
It says Opened pseduo zap
On Wed, Apr 02, 2008 at 02:16:56PM -0400, Jerry Geis wrote:
On Tue, Apr 01, 2008 at 01:44:39PM -0400, Jerry Geis wrote:
/ I have no card in this unit at this time.
// lsmod shows ztdummy loaded.
/
Just to make sure that this is not the problem, what's the output of:
zttest -c 3
--
Yes, some kernels don't work with ztdummy. This is discussed over and
over and over again on this mailing list. Check the archives.
Tzafrir Cohen wrote:
On Wed, Apr 02, 2008 at 02:16:56PM -0400, Jerry Geis wrote:
On Tue, Apr 01, 2008 at 01:44:39PM -0400, Jerry Geis wrote:
/ I have no card
Here I will say it http://xorcom.com
On Mon, Mar 31, 2008 at 6:01 PM, Al lists [EMAIL PROTECTED] wrote:
I'm looking to install a system with 80 FXS analog phones.
At this time the only cost effective solution is using a 4 port T1 card and
addit 600 channel bank.
Has anyone tried this
Hi,
maybe this has been asked before but I couldnt find a proper answer on
the web or list.
I want to use a analog V.92 modem to make outgoing (and possibly)
incoming phone call through a standard analog phone line.
I found on web it's easy been done via chan_modem.so module. But this
I haven't, didn't know if you knew off the top of your head.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales
Sent: Tuesday, April 01, 2008 7:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to
I'd say, save yourself the time and the frustration, drop the idea and
buy a real voice card.
Zoa
Ronny Forberger wrote:
Hi,
maybe this has been asked before but I couldnt find a proper answer on
the web or list.
I want to use a analog V.92 modem to make outgoing (and possibly)
You could also, conceivably, handle this outside of asterisk by using a
more complex MOH stream source. For instance, use a shoutcast client as
the MOH source, run your own shoutcast server streaming your music and
have a script set up to periodically interrupt the stream being served
to the
We are attempting to configure SIP trunking between asterisk 1.2.22 and a
Mitel 3300 CX box. The Mitel machine will gateway to the PSTN for us. I
found this earlier post about doing this from July:
http://lists.digium.com/pipermail/asterisk-users/2007-July/191630.html
Unfortunately the
On Wed, Apr 2, 2008 at 9:37 AM, Ruben Zamora [EMAIL PROTECTED] wrote:
I just want to know if anyone have problems with server DELL 1600,
Like: Hangup Call.
Give us some more details of your setup and you'll probably have
better chances of getting an answer.
-Erik
Solved. Firewall problem.
In case someone may run into this issue.
There is a firewall between Asterisk and the database. The firewall kills an
idle TCP connection after an hour. Asterisk and MySQL do not know this. Next
time a call comes in, Asterisk reuses the connection. To make use of the
Could you be like 1% more specific, with perhaps including any
relevant Errors, Log Files ?
What makes you think it is the DELL , not your T1 boards, or your
service provider or
Ruben Zamora wrote:
Hi
I just want to know if anyone have problems with server DELL 1600,
Like:
Greg Woods wrote:
Shaun Ruffell wrote:
svn co http://svn.digium.com/svn/zaptel/branches/[EMAIL PROTECTED]
zaptel-1.4-4122
Thank you, I will try that tonight when I get home and report back.
--Greg
___
-- Bandwidth and
Hello,
I am working on a project and have a few questions. I want to connect
one port of the TE205p to the PSTN, and another port to the PRI port
of a PBX. Basically Asterisk will sit between the PSTN and the
existing PBX. Are there any gotcha's I need to be aware of? Will the
existing PBX be
2008/4/2, Jean-Denis Girard [EMAIL PROTECTED]:
Development version of libpri (libpri-trunk) does include prliminary
support for BRI.
I took a look at :
http://svn.digium.com/view/libpri/trunk/
Though BRI support is mentioned several times but I couldn't find any
supported hardware list.
While not asterisk specific related - would be interesting to see
something similar done against straight voip calls.
You could maybe plot it against e.164.org polls but I think this would
be too small a sample set.
Can anyone think of something else Asterisk related we could plot this
against?
On Wed, 2008-04-02 at 21:18 +0200, Ronny Forberger wrote:
I want to use a analog V.92 modem to make outgoing (and possibly)
incoming phone call through a standard analog phone line.
When I asked this question, I was basically told that it isn't possible.
The problem is along the lines that
On Wed, 2008-04-02 at 15:23 -0500, Brent Davidson wrote:
the cords that ran between the wall jack and the jacks on the X100P
cords all ran between the server's 21 CRT monitor and the wall.
Not a problem here, as the monitor is on the other side of the room from
the server and the wire from
Hi list,
sorry if this has been discussed in the past, but I couldn't find anything
wise about it.
Since we had some trouble with the builtin hold function of some (all?) SNOM
320/360
phones, we decided to use the call parking feature in asterisk instead.
Assume, a call comes in with
Hi,
Has anyone information about BRI hardware supported by 1.6 libpri ?
In another thread, I was told a basic BRI card with HFC chipset (Bewan Gazel
128) was supported but I would delighted to lear about other harwarde (and
specifically about Digium B410P).
Regards
On Wed, 2008-04-02 at 13:26 -0700, Isaac McDonald wrote:
I am working on a project and have a few questions. I want to connect
one port of the TE205p to the PSTN, and another port to the PRI port
of a PBX. Basically Asterisk will sit between the PSTN and the
existing PBX.
This will work just
Olivier a écrit :
2008/4/2, Jean-Denis Girard [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]:
Development version of libpri (libpri-trunk) does include prliminary
support for BRI.
I took a look at :
http://svn.digium.com/view/libpri/trunk/
Though BRI support is mentioned
On Wed, Apr 02, 2008 at 11:17:05PM +0200, Olivier wrote:
Hi,
Has anyone information about BRI hardware supported by 1.6 libpri ?
In another thread, I was told a basic BRI card with HFC chipset (Bewan Gazel
128) was supported but I would delighted to lear about other harwarde (and
On Wed, Apr 2, 2008 at 11:05 PM, Brent Davidson
[EMAIL PROTECTED] wrote:
You could also, conceivably, handle this outside of asterisk by using a
more complex MOH stream source. For instance, use a shoutcast client as the
MOH source, run your own shoutcast server streaming your music and have
Well, not necessarily. You could have one MOH stream that gets
interrupted say every 10 seconds with a generic message. A caller that
gets connected to the MOH stream might come in in the middle of the
message, during music playback, or anywhere form 0-10 seconds before the
message plays
Its Nice, i agree, but we are looking at $4k to $5k with this.
On Wed, Apr 2, 2008 at 1:17 PM, Andrew Latham [EMAIL PROTECTED] wrote:
Here I will say it http://xorcom.com
On Mon, Mar 31, 2008 at 6:01 PM, Al lists [EMAIL PROTECTED] wrote:
I'm looking to install a system with 80 FXS
Hello all,
I;m having a (what seems to me) strange problem with some analog lines and
hangup detection.
The site I;m working on has 10 analog lines, my understanding is these are
broken up in 2 invidiaul hunt groups (no idea why, or if this is even true).
I;ve always been told that they
Yes, the Digium cards should support T1 CAS.
Andrew Latham wrote:
Here I will say it http://xorcom.com
On Mon, Mar 31, 2008 at 6:01 PM, Al lists [EMAIL PROTECTED] wrote:
I'm looking to install a system with 80 FXS analog phones.
At this time the only cost effective solution is using
Andrew Latham wrote:
Here I will say it http://xorcom.com
alternatively:
http://www.audiocodes.com/objects/30010_DS_MP-11X,%20MP-124D.pdf
On Mon, Mar 31, 2008 at 6:01 PM, Al lists [EMAIL PROTECTED] wrote:
I'm looking to install a system with 80 FXS analog phones.
At this time the only
Bad memories from AudioCodec :)
On Wed, Apr 2, 2008 at 7:48 PM, Edwin Lam [EMAIL PROTECTED]
wrote:
Andrew Latham wrote:
Here I will say it http://xorcom.com
alternatively:
http://www.audiocodes.com/objects/30010_DS_MP-11X,%20MP-124D.pdf
On Mon, Mar 31, 2008 at 6:01 PM, Al lists
Shaun Ruffell wrote:
If you have subversion installed on your server, could you try using
this version of zaptel:
http://svn.digium.com/svn/zaptel/branches/[EMAIL PROTECTED]
Thank you to everyone who offered assistance. For now, upgrading the
driver has fixed the problem: outbound
In the real world, just how good of recognition can you get based on
your experience ?
How much processing power do you find it takes ? I know that a dedicated
Voice Recognition for the
PC such as Dragon Naturally Speaking requires :
-a pretty beefy system
-that you use a limited set of
Hi Al,
I'm saying this politely so don't take it the wrong way. Go away and do
some research.
Learn the difference between NLVR (Dragon Dictate) and limited set
utterance recognition (Lumenvox).
Lumenvox is a great product for the price point and as their developer
kit is so reasonable you
And it's not going to happen now so give up dreaming about it (I have)
but search www.voip-info.org for TellMe and see what would be the
pinnacle for speech recognition for the Asterisk community.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357
+61-2-9016-5642
On Wed, Apr 2, 2008 at 10:24 PM, Al Baker [EMAIL PROTECTED] wrote:
Clearly all of this not feasible in a IVR environment, so, in the
absence of all this, just how good , and how sophisticated of a voice
recognition can one achieve ?
Have you ever called Google 411?
1-800-GOOG-411
It'll
Thank you.
I think that part of one's research is to ask in a user group what
peoples real word experience of a product has been. Particularly since
I was responding to a post by Philip in which he had said
So what is that you'd like to know?
Philipp
in response to a question of
I
Ok, I think that original request was answered more than a few days ago
by people who had actually used Lumenvox.
Re-reading your answer it appears you provided advice about a topic
where you hadn't actually utilized the product.
As several people who answered Phillip's original question -
Al lists wrote:
Bad memories from AudioCodec :)
Por que? I'm curious.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
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