${DIALSTATUS} to eventually define some
gotos because it's ANSWERED.
Any ideas as to what I should try?
Maybe change some setting in zapata.conf?
Thanks
Vieri
Shape Yahoo! in your own image. Join our
--- Noah Miller [EMAIL PROTECTED] wrote:
You can try using ChanIsAvail() to test beforehand
if the zap channels
will accept a call.
I'll try that. Thanks Noah.
My test was with disconnected analog lines.
I will also try to do the same but this time will keep
the line busy by placing a call
what to do in case the wires get
disconnected without having to change the config
files.
Thanks anyway.
Vieri
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--- Eric \ManxPower\ Wieling [EMAIL PROTECTED]
wrote:
You cannot detect disconnected analog lines in
Asterisk. You can't even
determine of the lines have dialtone. All you can
do is determine if
there is a current active asterisk managed call.
You're right, unfortunately. ;-(
If I
interfaces:
asterisk-chan_capi from
http://www.junghanns.net/asterisk/
chan-capi-cm from http://www.melware.org/ChanCapi
Can anyone please give me some hints as to loading the
isdn module? (or how to debug it)
Thanks
Vieri
--- Vieri [EMAIL PROTECTED] wrote:
Unable to load module chan_misdn.so
Never mind. I found out it was a permissions problem.
Either asterisk must be run as root or the asterisk
user must be part of the dialout group.
(tweaked /usr/sbin/amportal
Hi,
I would like to know if one can set the outgoing
caller ID within Asterisk when calls are going out
through:
1) an analog POTS line (I suppose not)
2) a telco BRI line (I don't think so)
3) a telco PRI line (maybe)
4) a voip provider (surely)
Thanks,
Vieri
this is simple to solve but I just can't find
the right option.
Has anyone seen this behavior in the GXW-4008 ATA or
similar?
Thanks,
Vieri
Moody friends. Drama queens. Your life? Nope! - their life, your story
--- Mr Shunz [EMAIL PROTECTED] wrote:
Caller ID Scheme as
ETSI-FSK Prior to Ringing with DTAS...
Thank you Daniele.
That seems to work.
I tested it on analog phones without a display.
I had previously experimented with different schemes
because I needed some of our phones to correctly
be
appreciated.
Thanks,
Vieri
Code snippet:
$ftime = time();
$fname = /tmp/asterisk_.$ftime..call;
$fname_call =
/var/spool/asterisk/outgoing/asterisk_.$ftime..call;
$fd = fopen($fname, 'w');
fwrite($fd, Channel: .$alerts.\n);
fwrite($fd, Callerid: IT 7021
--- Jerry Geis [EMAIL PROTECTED] wrote:
when executing a NOOP(caller id ${CALLERIDNUM})
I am using asterisk 1.2.17
I use CALLERID(num) or CALLERID(all) in 1.2+.
I don't know if that can help.
?
--- Vieri [EMAIL PROTECTED] wrote:
I am writing a cron script to check if certain
extensions are online and if they aren't then
Asterisk
creates a couple of .call files to notify another
set
of extensions or external numbers.
It works fine except for logging information.
What I'm doing
is something like
Zap/1-1 (for SIP I would get SIP/4053-082fc408 and
that allows me to extract the dialed extension
4053).
How can I get the dialed number when using a ZAP
channel?
(using asterisk 1.2.17)
Regards,
Vieri
PS: sorry if I dupe this post but I searched the
archives and I didn't find
--- Atis [EMAIL PROTECTED] wrote:
On 8/30/07, Vieri [EMAIL PROTECTED] wrote:
I am trying to retrieve the dialed peer number
but
it seems that ${DIALEDPEERNUMBER} is broken.
Also, I know that I could extract the dialed
number
from the ${CHANNEL} variable but this only works
for
SIP
--- Atis [EMAIL PROTECTED] wrote:
On 8/30/07, Vieri [EMAIL PROTECTED] wrote:
However, I'm still having some trouble trying to
understand why Asterisk does not log the ZAP
number as
is otherwise the case for SIP. A description of
this
problem is at:
http://lists.digium.com/pipermail
--- Atis [EMAIL PROTECTED] wrote:
You can always use different extension in
custom-NOTIFY, it doesn't
have to be s. It can be extension you are sending
call to or
anything else. And if this one is not enough, you
can set some
variables in call file and use them from dialplan
(and append
Hi,
I setup a queue (number 4050) with one static agent
(extension 4054).
What I would like is that when someone calls the 4050
queue and there are neither dynamic agents logged in
nor is the static agent 4054 on-line then the caller
gets out of the queue and falls into another context
(eg.
--- Mark Michelson [EMAIL PROTECTED] wrote:
Vieri wrote:
Hi,
I setup a queue (number 4050) with one static
agent
(extension 4054).
What I would like is that when someone calls the
4050
queue and there are neither dynamic agents
logged in
nor is the static agent 4054 on-line
-08311988 is busy
So how could I get the response code *without*
actually dialing from within an AGI script? (I don't
want to establish a call, just want to see if the SIP
client replies with a DND response code)
Like a ping of some sort...
Vieri
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
On Behalf Of Vieri
Sent: 13 September 2007 14:30
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] how to determine if a
SIP extension
has DND on oroff
I would like to determine through an AGI script
An Asterisk extension calls an Alcatel extension via a
PRI link which rings 4 times for about 10-15 seconds
and then drops.
So if the Alcatel user doesn't answer within 10-15
seconds the call is aborted.
(A timeout is *not* specified in the Asterisk Dial
command.)
It seems however that either
--- Steve Langstaff [EMAIL PROTECTED] wrote:
Can you hook into the qualify code somehow? - that
uses SIP OPTIONS.
I already knew of this wiki page:
http://www.voip-info.org/wiki/view/Asterisk+sip+qualify
So I did a sip show peer on the asterisk cli which I
am supposing is the same as the
a pri debug span X to see the actual
Q.931 ISDN messages that
are exchanged.
Vieri wrote:
An Asterisk extension calls an Alcatel extension
via a
PRI link which rings 4 times for about 10-15
seconds
and then drops.
So if the Alcatel user doesn't answer within 10-15
seconds the call
--- Eric \ManxPower\ Wieling [EMAIL PROTECTED]
wrote:
Looks like the Alcatel is sending back a busy.
Note:
I'm using libpri patched with BRIstuff.
http://ftp.digium.com/pub/libpri/libpri-1.2.4.tar.gz
http://www.junghanns.net/downloads/bristuff-0.3.0-PRE-1y-d.tar.gz
--- Steve Langstaff [EMAIL PROTECTED] wrote:
I don't know about the 1.4 source, but in 1.2 I
guess you would have to
add some more code to
handle_response_peerpoke()
to handle the case where you got a 486 response from
the peer.
ok thanks, so that just seems to confirm that Asterisk
--- Steve Langstaff [EMAIL PROTECTED] wrote:
The OP was asking whether they could update
Asterisk's DND status
I think that they *actually* want to do some queue
management based on
the DND button of the (SIP) phone.
-Original Message-
From: [EMAIL PROTECTED]
and which side
is actually producing this event?
Thanks,
Vieri
Luggage? GPS? Comic books?
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but I don't know how.
Vieri
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on the Asterisk side?
-Vieri
Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news,
photos more.
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some kind of
pulse and unless I change adequately busypattern
then it considers it as a BUSY signal.
Since I have no idea of what the busypattern should be
then I'll just try to switch off busydetect which
should not be needed on a PRI.
Many thanks.
Vieri
--- Eric \ManxPower\ Wieling [EMAIL PROTECTED]
wrote:
Set callprogress=no and busydetect=no in
/etc/asterisk/zapata.conf.
Don't use them. They cause random disconnects.
Thanks, will try.
have core dumps in /tmp.
What can I do to isolate the cause of these
segmentation faults?
Thank you for your advice,
Vieri
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table info available.
(gdb)
(gdb) thread apply all bt
... not posting because too long here ...
Thanks,
Vieri
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--- Atis Lezdins [EMAIL PROTECTED] wrote:
We also experienced this problem on 1.2, but i'm not
sure that this is
registered in bug database. You should check
bugs.digium.com and if it's
still valid for 1.4, you should post your backtraces
there.
Actually, I'm using 1.2.21.1 so since 1.2
Hi,
I would like to know if the following is possible:
* how to accept only one call at a time on a
particular SIP extension (softphone). I'm referring to
incoming calls. Can it be done on the server side or
just on the client? ie. all other incoming calls will
just be dropped while the
I am evaluating the best way to make a high avail and
load balanced system.
I have two identical asterisk servers. Most clients
are SIP phones. The only special hardware I have on
both systems (they are identical) is: 1 E1 PRI card
and 1 4-port BRI card.
I have 8 ISDN lines so 4 go to each pbx
Are there any scripts to convert * .conf files to SQL
for Realtime?
Looking for last minute shopping deals?
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Hi.
Is it possible to override the standard DB function in
Asterisk?
My dialplan contains a lot of calls to Set(DB(...))
and ${DB(...)} which of course use astdb to
store/read data. I would like to stop using astdb and
switch to Clustered MySQL (I don't suppose clustered
astdb exists?).
So
the _X. pattern match is wrong?
How can I match an alphanumeric string?
Thanks,
Vieri
Be a better friend, newshound, and
know-it-all with Yahoo! Mobile. Try it now.
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--- Vieri [EMAIL PROTECTED] wrote:
On peer2:
[dundi-incoming]
exten = _X.,1,NoOp(Received EXTEN ${EXTEN}.)
exten = _X.,n,Set(EXTTODIAL=${CUT(EXTEN|#|1)})
exten = _X.,n,Set(DUNDIVAR=${CUT(EXTEN|#|2)})
exten = _X.,1,NoOp(Extracted extension ${EXTTODIAL}
and DUNDi variable ${DUNDIVAR
--- Richard Lyman [EMAIL PROTECTED] wrote:
Vieri wrote:
Hi.
I am trying to pass a variable from one Asterisk
PBX
to another.
I'm using DUNDi with IAX2. Is there a way to do
it?
I tried the following but it fails.
On peer1:
[dundi-outgoing]
switch = DUNDI/priv
What is the format of the UNIQUEID variable?
It seems to be something like:
40651204817492.56
Does it always have the pattern
long_number.short_number?
Be a better friend, newshound, and
know-it-all
] On
Behalf Of Vieri
Sent: Thursday, 6 March 2008 2:57 AM
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] Passing variables
between two DUNDi/IAX2 peers
--- Richard Lyman [EMAIL PROTECTED] wrote:
Vieri wrote:
Hi.
I am trying to pass a variable
--- Tony Mountifield [EMAIL PROTECTED] wrote:
In Asterisk 1.4 or later, an optional system name
can be defined in
asterisk.conf, and if defined, the unique ID
becomes:
system_name-timestamp.seq_num
Thanks!
So for the sake of backward compatibility, if I dont'
define sysname in 1.4 then
I've been searching the Internet for information
regarding the replacement of astdb with a modern sql
engine.
There are several reasons one would like to do this.
First of all, external applications have a hard time
reading/writing to the now-old astdb format.
Also (and this is what interests me
--- Bruce Reeves [EMAIL PROTECTED] wrote:
Vieri,
What values are you looking to move from astdb?
I have used realtime to store values for call
features and other
functions in the dial plan. I'm curious what you are
looking to do.
Thanks for the feedback Bruce.
What I'm trying to do
--- Atis Lezdins [EMAIL PROTECTED] wrote:
For your own custom data you can use Realtime engine
- it has INSERT
and DELETE support in 1.6, and it's easily
backportable to 1.4 (if
you're interested i can give you working patches).
All you have to do
is declare realtime class in
Would it be possible to modify the API calls that are
currently going to the AstDB code within Asterisk, and
put a translation layer to have them use the func_odbc
instead (or either one)?
I came across some old code of an odbc version of
app_db which I suppose is obsolete.
--- Vieri [EMAIL PROTECTED] wrote:
Would it be possible to modify the API calls that
are
currently going to the AstDB code within Asterisk,
and
put a translation layer to have them use the
func_odbc
instead (or either one)?
At a lower level, for everything Asterisk does to
its
AstDB
According to
http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS
when a caller hangs up before the callee has time to
pick the phone up then DIALSTATUS should be CANCEL.
And it is.
However, the disposition field in the CDR table is NO
ANSWER.
So if I analyze the CDR data I won't be able
--- Vieri [EMAIL PROTECTED] wrote:
According to
http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS
when a caller hangs up before the callee has time to
pick the phone up then DIALSTATUS should be CANCEL.
And it is.
However, the disposition field in the CDR table is
NO
ANSWER
Hi,
Surely, I must be overlooking something. If I run the
following SQL queries I don't get the same number of
rows. Is this coherent?
mysql select * from queue_log where queuename =
'4010' and FROM_UNIXTIME(time) between 2008030800
and 20080313145900 group by callid;
357 rows in set (0.01
--- Ex Vito [EMAIL PROTECTED] wrote:
...as long as the destination does not answer
you'll get
a NO ANSWER disposition.
So, if in your case you want to know if a user
answered
the phone, then, yes, you will have to add the
DIALSTATUS
value to the CDR, probably in the CDR's
--- Atis Lezdins [EMAIL PROTECTED] wrote:
Hmm, didn't knew that queue_log can be written into
MySQL..
Asterisk 1.6 beta has that through realtime.
But I'm using a custom import script in earlier
Asterisk versions.
Is callid in queue_log the same uniqueid?
yes.
You can do
something like
I set uniquename = MYHOST in asterisk.conf (under
[options]) so that my uniqueid data shows up as
MYHOST.time.seq.
First of all, I would like to know if uniquename (or
sysname?) will still be valid across future * versions
(mainly 1.6).
Secondly, is there a way to specify uniquename as an
--- Matt Riddell [EMAIL PROTECTED] wrote:
http://bugs.digium.com/view.php?id=12230
Thanks Matt.
However, I may be wrong but this isn't exactly what
I'm looking for. I would like Asterisk to
transparently set my CDR(disposition) field to
reflect if a call has simply timed out (NO ANSWER) or
if
--- Vieri [EMAIL PROTECTED] wrote:
I set uniquename = MYHOST in asterisk.conf (under
[options]) so that my uniqueid data shows up as
MYHOST.time.seq.
First of all, I would like to know if uniquename (or
sysname?) will still be valid across future *
versions
(mainly 1.6).
Secondly
How can I force soft hangup (if that makes sense)?
show channels reveals a stale sip channel. It's of
an analog phone behind a Grandstream ATA which was
communicating with another SIP softphone. The latter
crashed. A soft hangup of the softphone seems to have
worked but it doesn't for the
--- Steve Davies [EMAIL PROTECTED] wrote:
On 25/03/2008, Gordon Henderson
[EMAIL PROTECTED] wrote:
On Tue, 25 Mar 2008, Vieri wrote:
How can I force soft hangup (if that makes
sense)?
show channels reveals a stale sip channel.
It's of
an analog phone behind a Grandstream
--- Steve Davies [EMAIL PROTECTED] wrote:
Using rtptimeout and rtpholdtimeout settings in
sip.conf
I set
rtptimeout=10
rtpholdtimeout=30
(just for testing; I know these values are way too
low)
then did a
CLI sip reload
and waited more than 30 seconds.
The SIP channel is still there (InUse).
--- Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Tuesday 25 March 2008 10:17:54 Vieri wrote:
--- Steve Davies [EMAIL PROTECTED] wrote:
Using rtptimeout and rtpholdtimeout settings in
sip.conf
I set
rtptimeout=10
rtpholdtimeout=30
(just for testing; I know these values are way
--- Tilghman Lesher
[EMAIL PROTECTED] wrote:
The SIP channel is still there (InUse).
Channel Location State
Application(Data)
SIP/6010-b38d53e0[EMAIL PROTECTED]:8 Up
Dial(SIP/4053||tTwW)
Should I interpret the above that it's in an
infinite
an agi command such as
QueueRemove(busyagent...). When the agent is free
again I suppose the same event is triggered and the
custom script can QueueAdd(freeagent...).
Could anyone please give me some pointers on this?
Thanks!
Vieri
for
call-limit=1)?
Thanks,
Vieri
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--- Atis Lezdins [EMAIL PROTECTED] wrote:
On Thu, Mar 27, 2008 at 6:32 PM, Vieri
[EMAIL PROTECTED] wrote:
I have a queue I configured as strict and a cron
script I use to QueueAdd and QueueRemove agents
according to my company's requirements. Usually I
have
2 or 3 agents at a time
--- Johansson Olle E [EMAIL PROTECTED] wrote:
Remember that if you enable call-limit=1 with a
type=friend, you will
actually have one inbound call (on the user)
and one outbound call (on the peer).
Groupcount in the dialplan is propably a better
solution to enforce
call limits than
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)
--- 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
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
Zaptel seems to compile fine until I enter xpp/utils
and make there.
I get:
xpp/utils # make
cc -o print_modes -g -Wall print_modes.c
print_modes.c: In function `main':
print_modes.c:9: error: `fxo_modes' undeclared (first
use in this function)
print_modes.c:9: error: (Each undeclared identifier
.
Compiles ok.
Vieri
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I already have ilbc installed on my system. The files
are:
/usr/include/ilbc/iLBC_decode.h
/usr/include/ilbc/iLBC_define.h
/usr/include/ilbc/iLBC_encode.h
/usr/lib/libilbc.a
/usr/lib/libilbc.la
/usr/lib/libilbc.so - libilbc.so.0.0.0
/usr/lib/libilbc.so.0 - libilbc.so.0.0.0
--- Kevin P. Fleming [EMAIL PROTECTED] wrote:
Vieri wrote:
How can I tell the make system in 1.4.19 that ilbc
is
already on the system and that it should link to
/usr/lib/libilbc.a?
Shouldn't the configure script do that?
No; the Asterisk build system has never had support
--- Nestor A. Diaz [EMAIL PROTECTED] wrote:
Mojo with Horan Company, LLC wrote:
Nestor A. Diaz wrote:
1. I use a queue with just on sip device, one
call at a time, however
and without reason just after some couple of
hours the sip device show
in use and then no calls are
--- Kevin P. Fleming [EMAIL PROTECTED] wrote:
Vieri wrote:
So basically I'm wondering if the Asterisk
make/configure process could do steps 1 and 2
automagically for me.
I can't find any other Linux distribution that
provides libilbc, so this
would be a very Gentoo-specific change
--- Nestor A. Diaz [EMAIL PROTECTED] wrote:
Vieri wrote:
Did you try a show channels to see if there were
stale channels for peer 200?
I had the same problem you describe but it was due
to
hung channels (used * 1.4.18.1 with rtp*timeout
and
saw inuse peers during the pre-timeout
--- Nestor A. Diaz [EMAIL PROTECTED] wrote:
ok, thanks, does rtp*timeout work if i have
canreinvite=yes ? since rtp
traffic is not passing thought asterisk, or i have
to put canreinvite=no ?
In my setup it doesn't really matter since calls are
coming in through PSTN-IVR-QUEUE-SIP
--- bee-beeep [EMAIL PROTECTED] wrote:
It works fine in every case, with disabling transfer
in Dial() options
2008/4/25 Grey Man [EMAIL PROTECTED]:
Thanks to your answers, but i found more
beautiful way to do this -
there is some system variable
__TRANSFER_CONTEXT, which
channel and the wait time. I
would require correlating the data to the caller's ID.
Has anyone already done something similar?
A simple example/script/suggestion would be greatly
appreciated.
Thanks,
Vieri
--- Atis Lezdins [EMAIL PROTECTED] wrote:
On Mon, Apr 28, 2008 at 8:34 PM, Vieri
[EMAIL PROTECTED] wrote:
How can I get a list of the callers within a
specific
queue at any given moment?
I need to get the caller IDs of all active calls
in a
queue then send them out via a udp
it. However, I think that your patch
should hit SVN and I wouldn't mind testing it.
Thanks,
Vieri
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--- Atis Lezdins [EMAIL PROTECTED] wrote:
So, the issue is
http://bugs.digium.com/view.php?id=12556, feel free
to comment about usage.
I also posted backport to 1.4.19 at
http://ftp.iq-labs.net/realtime_queue_callers-1.4/
but for this You
will need to also apply backport for realtime
,
10 callers in queue 1000 on pbx1 and the 11th call
arrives on pbx2 with position 1.
Is there a way of coherently setting up a clustered
queue?
Does anyone have examples/workarounds/links?
Thanks!
Vieri
--- Vieri [EMAIL PROTECTED] wrote:
Is there a way of coherently setting up a clustered
queue?
Does anyone have examples/workarounds/links?
I guess there's no easy (open-source) solution to this
problem, at least for now (* 1.6?).
I believe Yate2 has something on this but it's still
alpha
--- Vieri [EMAIL PROTECTED] wrote:
Is there a way of coherently setting up a clustered
queue?
Does anyone have examples/workarounds/links?
I guess this isn't easy to implement, at least in
current Asterisk versions (* 1.6?).
I think Yate2 may have support for clustered queues
but it's still
Hi,
I'm having trouble connecting two Asterisk boxes via a IAX2 friend trunk.
iax2 show peers on both boxes seem to show that all's fine (Status OK on
qualify=yes peer).
voip1 is an Asterisk 1.2.27 production server.
voip2 is an Asterisk 1.4.21 experimental server in the same gigabit LAN.
If I
from an extension on one box to another extension on
another box (or simply DIAL a IAX2 friend trunk between the boxes).
How would I go about capturing a call ringing on ext1 from ext2 with *8 or
similar knowing that ext1 is registered to box1 and ext2 to box2?
Thanks,
Vieri
\ /
Clustered Realtime
| |
SIP members Queues
Is Asterisk (1.2 and 1.4) ready for clustered realtime?
Vieri
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(the ATAs will be
able to connect to the same Asterisk server but through the non-failing switch).
Linksys has a similar 8-FXS ATA but it only has one ethernet interface.
Thanks,
Vieri
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--- On Mon, 7/7/08, Mark Michelson [EMAIL PROTECTED] wrote:
From: Mark Michelson [EMAIL PROTECTED]
Subject: Re: [asterisk-users] queue member state
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Monday, July 7, 2008, 10:54 AM
There is a
? The ones I've seen only had a
LAN port.
Also, what's your experience with it?
Do you only have one?
Has it been stable?
Thanks
Vieri
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AstriCon 2008 - September 22 - 25 Phoenix
= asteriskcluster
Option = 3
(isql astdb_cluster works fine even in * 1.2.27 but asterisk -rx odbc
show doesn't)
Thanks
Vieri
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AstriCon 2008 - September 22
Replying to myself.
A reload isn't enough in 1.2.27. I needed to restart asterisk.
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AstriCon 2008 - September 22 - 25 Phoenix, Arizona
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into account. The extensions will
stay on box 2 and will move to box 1 only if:
- box 2 dies
- or I wait around 30 minutes (I don't what this timeout could be)
I've tried it on Asterisk 1.4.21.2 and 1.2.30.
Any ideas?
Thanks,
Vieri
anything else about it (how
to connect, how to change config, how to reset the device, etc). There's
absolutely nothing regarding RS-232.
If someone has this or a similar device and accessed it via serial port then
I'd greatly appreciate some quick tips.
Thanks,
Vieri
Never mind. I set the wrong baud rate. The right values are 115200-8-N-1-No
flow control. However, the serial connection is as good or as useless as the
telent connection. I have no way to restore factory settings.
--- On Tue, 8/5/08, Vieri [EMAIL PROTECTED] wrote:
I realize this may
with a complete list of telnet commands.
Vieri
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Thanks,
Vieri
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(non-ATA), we would need
to buy more ethernet switches (currently they're all full) and tunnel cables
thtough ceilings and walls. In other words, it would cost a lot more than to
simply buy ATAs. What I'm looking for however are STABLE, RELIABLE ATAs...
Thanks for the feedback,
Vieri
Thanks for the feedback.
I'm particularly curious to know if anyone has tried a TDMoE channel bank.
Spidermux seems to be one of the few vendors available. It's the closest I can
get to an ATA-like device (ie. no special hardware, just ethernet) and it
also offers an easy failover mechanism to
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