[EMAIL PROTECTED] wrote on 11/19/2005
02:04:54 AM:
> Have you tried the new gain (g) option for the voicemail application?
> From 'show application voicemail':
>
> g(#) - Use the specified amount of gain when recording the voicemail
> message. The units are whole-number
decibels (d
Hello!
In honor of 1.2 being released, and
now that I'm in the mindset to go spelunking into Asterisk code to address
minor annoyances, I have a second issue:
Every voicemail system I'm aware of
(my Sprint cellphone voicemail, Nortel systems, InterTel telehpone sytsems
and others that slip my mi
[EMAIL PROTECTED] wrote on 11/16/2005
09:46:17 PM:
> Hi,
> Yes, I'm using wav for my
recording and the file is quite large.
I too am using WAV files because of the volume issue:
WAV files are shifted two bits louder than any other format. (slight
details here: http://lists.digium.co
[EMAIL PROTECTED] wrote on 11/15/2005
06:42:40 PM:
> That being said, and as I mentioned earlier, your cheapest choice
is
> to go to eBay and search for X100P.
Here's a question: why are you building your
hobby box? To gain practical experience? Then forget the X100:
it's like learning Win
[EMAIL PROTECTED] wrote on 11/15/2005
02:53:54 PM:
> Ending last year I used\sold several hundred of product#: FM-INL92SW.
>
> Google for it...you'll find some for cheap.
Along those lines: are there drivers for the
X100/X101 that allow it to act as a normal v.92 modem, even if it's just
under
[EMAIL PROTECTED] wrote on 11/15/2005
05:42:44 AM:
> Olle E. Johansson wrote:
> > be seen as a sample of a full prompt set and something that is
extremely
>
> This actually leads to a question I've had for a while: Is there a
list
> somewhere of all the prompts (by filename) and what is said? I
[EMAIL PROTECTED] wrote on 11/07/2005
01:17:31 PM:
> HW: HP DL360 1GB Ram Single 3GHz Pentium (with Hyper-threading turned
off).
> OS: CentOS 4.2
> Dual Embedded NIC enabled
> USB disabled
> serial disabled
> printer disabled
> 2x73GB SCSI in HW Raid 1
>
> What is the opinion of this fine list
[EMAIL PROTECTED] wrote on 11/03/2005
11:49:12 AM:
> Use 'timestamp=yes' in asterisk.conf instead of -T.
This is exactly what I was looking for. Thank
you very much!
Tim Massey
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Use
[EMAIL PROTECTED] wrote on 11/04/2005
04:34:18 PM:
> Try calleveryone.com Yes.. I have blown their trumpet before.
They
> are a very good company with great support.
Do they support IAX or just SIP? I've been reluctant
to use a SIP provider for a number of reasons, including difficulties in
[EMAIL PROTECTED] wrote on 11/03/2005
09:03:44 PM:
>
> > > Might be worth it to read the stuff in /usr/src/asterisk/doc
and in
> > > particular the README.asterisk.conf file. Lots of other
good stuff
> > > in that directory as well. (Not much need to read the source
now.)
> >
> > Thank you. I
[EMAIL PROTECTED] wrote on 11/03/2005
03:33:06 PM:
> Might be worth it to read the stuff in /usr/src/asterisk/doc and in
> particular the README.asterisk.conf file. Lots of other good stuff
> in that directory as well. (Not much need to read the source now.)
Thank you. I don't mind being told t
[EMAIL PROTECTED] wrote on 11/01/2005
09:05:22 PM:
>
> Hello!
>
> A couple of weeks ago I mentioned echo issues I was having. It
> turns out that the echo was only happening for the first 30 seconds
> or so, so the echo cancellers *were* working, just not training
> well. I wish they had
[EMAIL PROTECTED] wrote on 11/03/2005
11:53:17 AM:
> Chris Wade wrote:
> > Use 'timestamp=yes' in asterisk.conf instead of -T.
> >
> > -T only affects messages generated by THIS connection (ie asterisk
-RT
> > generated messages... not server generated messages.
> >
> > 'timestamp=yes' affects
Hello!
A couple of weeks ago I mentioned echo
issues I was having. It turns out that the echo was only happening
for the first 30 seconds or so, so the echo cancellers *were* working,
just not training well. I wish they had told me that 2 weeks ago!
So, over the weekend, I made two changes:
1
[EMAIL PROTECTED] wrote on 10/31/2005
02:08:24 PM:
> Does anyone have any Motherboards to recommend us?
> Any part numbers for Celeron or P4 ?
For the record, I've found that kernel version has
a lot to do with it, too. CentOS 3.4 gives us 100% zttests. CentOS
4.2 gives us 99.9% tests.
Mother
[EMAIL PROTECTED] wrote on 10/31/2005
08:53:35 AM:
> On Monday 31 October 2005 09:30, [EMAIL PROTECTED] wrote:
> > Is there a way to add timestamps to each line in the console
so you know
> > exactly how long a call took? Or is there another way of
telling directly
> > within the console?
>
> O
Hello!
Lately, I've been keeping a close eye on an Asterisk box by staying logged
into the console for long periods of time. However, it can be very
difficult to know how long a telephone call lasts when this is all you
see:
-- Executing Dial("SIP/SIP105-8e34",
"Zap/g2/|60|t") in new stack
[EMAIL PROTECTED] wrote on 10/29/2005
04:01:26 PM:
> If I add this symbolic link creation into the statup scripts then
like I
> said zaptel working fine, however this is obviously not the right
way to
> fix this issue.
Are you doing "make config" when you compile
Zaptel? It does all of this f
[EMAIL PROTECTED] wrote on 10/27/2005
03:11:11 PM:
> > Are these settings persistent across reboots? The README
for
> fxotune seems to mention that you
> > need to do a "fxotune -s" in order to reload the card
with the
> analyzed settings (rather than
> > take the 20 minutes it seems to take o
[EMAIL PROTECTED] wrote on 10/27/2005
08:22:04 AM:
> > If you do an fxotune and all of the coefficients are 0, does
this
> mean that fxotune is not
> making
> > any changes?
>
> Based on what Matt has mentioned previously, fxotune only sets the
impedence
> to proper values today. He has not im
[EMAIL PROTECTED] wrote on 10/26/2005
05:09:30 PM:
> On Oct 26, 2005, at 3:40 PM, Mark Quitoriano wrote:
>
> > Hi list, i'm having a problem with asterisk+pstn termination,
i just
> > bought a TDM400p and connect my phone line(bellsouth) now when
im
> > using the pstn through asterisk there's
[EMAIL PROTECTED] wrote on 10/24/2005
08:09:27 PM:
> Also, if you're not already, try using the kb1 echo canceller from
> CVS-HEAD without aggressive cancellation before taking time to do
any of
> the above. It can be dropped into stable if needed by just copying
it
> (and the contents of the hea
Hello!
I have a question regarding time-based
includes in the dialplan. How are boundary conditions handled? And
is there a definitive, documented procedure for how to handle overlapping
time includes? For example, if I want to have day/night service from
8 A.M. to 5 P.M., there are two ways I
Hello!
Just thought I'd let everyone know that
a new revision has popped out from Digium: Rev J. I don't have
an I board in front of me to compare with, so I can't tell you what's different
(besides a bunch more text on the back). It looks like there is a
PE-68624 chip near each RJ-45 connecto
[EMAIL PROTECTED] wrote on 10/17/2005
12:45:13 PM:
> > > Here's a couple of ways to determine levels...
> > >
> > > 1. using the model 4 transmission test set, attach the tone
generator
> > > to one analog pstn line and the transmission level test
jacks to a
> > > second pstn line. Dial from one
[EMAIL PROTECTED] wrote on 10/16/2005
07:49:38 AM:
> Here's a couple of ways to determine levels...
>
> 1. using the model 4 transmission test set, attach the tone generator
> to one analog pstn line and the transmission level test jacks to a
> second pstn line. Dial from one line to other and m
[EMAIL PROTECTED] wrote on 10/12/2005
01:23:57 PM:
> On Wed, Oct 12, 2005 at 12:05:32PM -0400, [EMAIL PROTECTED] wrote:
> > I am in the middle of trying to get a milliwatt test line to
calibrate the
> > rxgain properly. However, this won't help me with the txgain,
will it?
> > How can I proper
Hello!
I'm having an echo problem with a TDM
card. The TDM card is being fed by a channel bank just 12 or so feet
away. When you put an analog handset on the line, both the RX and
TX volume seem to be just fine. However, when I use the TDM card,
I have to have an rxgain of 13.5, and even then,
Hello!
While performing some zttest's for some
time today, I was also keeping an eye of a top of the machine. While
the zttest was running, I also had a ssh-keygen and a dd creating a 5GB
file on an EXT3 partition running. I noticed that for the most part,
I got a decent number of 100%'s, and a
Hello!
A small suggestion for an improvement
to zttest: some sort of histogram to show a broader range of the
results that are being returned. For example, on a test machine I
ran each of the following items in separate infinite loops at the same
time:
ssh-keygen -b 8192 -t rsa -f /test.key
dd
[EMAIL PROTECTED] wrote on 10/01/2005
10:17:47 AM:
> Is there a way with either RHEL or CentOS to force it to use an
> APIC-enabled kernel? I've tried Googling but no success.
I can find no way of doing this during the install.
If you have a single processor system, AFAIK you are stuck with
[EMAIL PROTECTED] wrote on 09/30/2005
01:10:34 PM:
> On Fri, Sep 30, 2005 at 01:32:07PM +0100, Angus Comber wrote:
> > Hello
> >
> > I am using a VIA Epia ME6000 with a 600MHz Eden Fanless CPU.
Is this
> > likely to be enough power for a 8 extension system with 6 external
pstn
> > lines?
>
>
Hello!
I'm setting up Asterisk on a new system. In the past, all of my Asterisk
boxes have either been embedded-style systems that do not supoort APIC,
or multi-processor systems where APIC comes along with SMP. However,
now I'm trying to install Asterisk on a single CPU (and non-HT) system
tha
[EMAIL PROTECTED] wrote on 09/27/2005
01:18:35 PM:
>
> Hi I have looked around but I cant find an answer for this,
> I randomly get the error 'TDM PCI Master abort' and the system locks
up.
> All I have found so far are a couple other posts on it but no solution.
> Running fedora core 3, asteris
[EMAIL PROTECTED] wrote on 09/27/2005
03:13:21 AM:
> Hi,
>
> I did dmesg | tail it says ...
>
dmesg | tail
> f6 != 58
> f7 != 59
> f8 != 58
> f9 != 59
> fa != 58
> fb != 59
> fc != 58
> fd != 59
> fe != 58
> Freshmaker failed register test
>
Hello!
Given the current discusison regarding ztmonitor, line testing, etc., I've
been looking into purchasing a used transmission test set. From my
research, it seems that there are two items that might fit the bill: the
HP 3551A and the HP 4935A.
I know nothing about these specific devices
[EMAIL PROTECTED] wrote on 07/18/2005 11:56:06 AM:
>
> > Recently, I installed TDM04B 4 FXO card on to my Asterisk box and
> > installation went perfect.
> >
> > The only problem I am facing is the Voice mail has very poor quality
> > when any users leave voice message via PSTN line.
> >
> >
[EMAIL PROTECTED] wrote on 06/28/2005 07:52:50 AM:
> I was able to raise the volume from inaudible to acceptable by
> increasing the RxGain in zapata.conf by 5db. I'd rather not go the
> uncomressed wav route, as it will chew up storage in my email system.
I know I'm way behind on reading this,
[EMAIL PROTECTED] wrote on 04/19/2005 01:32:57 AM:
> ** Extract begins **
>
> SCSI RAID can cause the problem. If disabling hyper threading does not
> resolve your problem my next suggest would be to revert to a PATA IDE
> hard drive solution configured to UDMA level 2 using hdparm. SCSI or
[EMAIL PROTECTED] wrote on 04/17/2005 11:02:51 AM:
> On April 17, 2005 05:55 am, Tom Fanning wrote:
> > Illegal instruction (core dumped)
>
> Sounds like you have compiled asterisk for a processor that is "greater"
than
> the processor you're running on. I.e. compiled and told it to use P4
>
[EMAIL PROTECTED] wrote on 04/15/2005 01:45:22 AM:
> Digium have told us that a problem that we are having (with accuracy of
> zap interface as measured using zttest) may be due to the fact that we
> have a Xeon processor with hyperthreading and have suggested turning H/T
> off.
>
I've never
[EMAIL PROTECTED] wrote on 04/12/2005 11:36:47 PM:
> [EMAIL PROTECTED] wrote:
>
> > In other words, a PCI-based co-processor would double the PCI bus
> > bandwidth necessary. And with a latency-sensitive product like voice,
bus
> > contention is not something you want to add to! :)
>
> It o
[EMAIL PROTECTED] wrote on 04/12/2005 10:51:49 AM:
> Andrew Kohlsmith wrote:
>
> > secondary card for DSP functions is very inefficient of the PCI
> bus. I'd be
> > curious to know if the Digium cards can even do PCI-PCI DMA.
>
> The Digium TDM cards can DMA into any RAM accessible over the P
[EMAIL PROTECTED] wrote on 04/01/2005 03:24:49 PM:
> Try adding the module parameter boostringer=1 when loading the wctdm
> driver. This raises the ringing volts to 89V peak.
Is there a list of these anywhere? This is now the third one I've heard
of, with no documentation: lowpower (IIRC), ro
Paul wrote:
>I'd like to setup my Asterisk box to receive a call on the incoming POTS
>line and immediately redirect back out to connect to another phone
number.
>Im thinking I could use either the threeway feature of that POTS line, or
a
>second POTS connected to a different FXO card. Does ANYO
[EMAIL PROTECTED] wrote on 04/01/2005 09:04:38 AM:
>
> Take a look at the voicemail.conf.sample that comes with asterisk.
> Inside you will see how to change the voicemail email message that is
> cerated and add the phone number (and remove the name) for callerid.
Thanks. Once I found that it w
[EMAIL PROTECTED] wrote on 04/01/2005 12:36:07 AM:
> I'm new to the VOIP world and need some advice. I currently have a
> premium/ full functioned Panasonic PBX installed in my house/ small
> office... and have some extra unused telco lines available on the
> PBX. I'd like to use one of these
[EMAIL PROTECTED] wrote on 04/01/2005 12:03:15 AM:
> How do you get it to say where its from in the first place? ;-)
It just does! :) I've never done anything to enable it: It just happens
automatically.
For clarification, this is in an e-mail sent to me when I receive a
voicemail. This i
Hello!
When someone calls into my toll-free number delivered via IAX, the
caller's number shows up on my SIP phone. However, when I receive an
e-mail voicemail message, I get this message:
Just wanted to let you know you were just left a 0:01 long message
(number 2)
in mailbox 200 from Toll-
[EMAIL PROTECTED] wrote on 03/31/2005 02:24:11 PM:
> > -Original Message-
> > From: Rich Adamson [mailto:[EMAIL PROTECTED]
>
> > Its an odd thing. Some people have to reload, others don't, and there
> > has been no effort to determine why it occurs. I've got two systems
> > that do have t
[EMAIL PROTECTED] wrote on 03/30/2005 01:46:10 PM:
> Looking for reccomendations for a physically small box configuration
> that will do:
> Run Asterisk
> One T1 Card
> One LAN port
> Enough CPU power to handle encoding/decoding all 24 T1 channels
> to/from G.729a
>
> Someone men
[EMAIL PROTECTED] wrote on 03/28/2005 11:50:25 PM:
> Why does the agent has to be always "connect"? Is there a way to
> close the connection and have * to call the correct agent when a call
arrives?
If you want this to work through NAT, the soft clients will have to keep a
connection open. Th
[EMAIL PROTECTED] wrote on 03/28/2005 03:24:50 PM:
> [EMAIL PROTECTED] wrote on 03/28/2005 01:19:03 PM:
> > ; logn distance calls
> > exten => _91NXXNXX,1,NoOp("Dialing:
"${TRUNK}/${EXTEN:${TRUNKMSD1}})
> > exten => _91NXXNXX,2,Dial(${TRUNK}/${EXTEN:${TRUNKMSD1}})
> > exten => _91NXXNXXX
[EMAIL PROTECTED] wrote on 03/28/2005 01:19:03 PM:
>
> TRUNKMSD1=1 ; MSD digits to strip
> (usually 1 or 0)
> TRUNKMSD2=2 ; MSD digits to strip
> (usually 1 or 0)
>
> ; logn distance calls
> exten => _91NXXNXX,1,NoOp("Di
Steven Critchfield <[EMAIL PROTECTED]> wrote on 03/28/2005 11:44:03 AM:
> Depends on what functions you are trying to implement. Hold isn't hard
> on a regular phone. Transfer isn't hard. Voicemail access isn't hard.
> Beyond that, there isn't a lot that needs to be done.
>
> If you find that yo
Hello!
Now that I finally have my TDM board working, I want to move forward with
using PBX functions. However, it seems cumbersome to use standard POTS
telephones with Asterisk. I know that there are many of you installing
even large systems based on channel banks and analog telephones. What
[EMAIL PROTECTED] wrote on 03/25/2005 09:14:42 AM:
> Hello
>
> I want to to know if the motherboards VIA are fully supporte by
asterisk.
This is a complex question. The *software* runs on Mini-ITX (what I
assume you're asking about) just fine. The *hardware* *may* have issues
however.
Thes
[EMAIL PROTECTED] wrote on 03/25/2005 09:14:42 AM:
> Hello
>
> I want to to know if the motherboards VIA are fully supporte by
asterisk.
This is a complex question.
The *software* is fully supported. Depending on the CPU you use, you may
have to modify the makefiles (some VIA CPU's do not im
[EMAIL PROTECTED] wrote on 03/24/2005 06:42:24 PM:
> Does anyone know how to qualify existing Cat3 wiring for use as a LAN?
That's easy: Cat3 is able to handle 10Mbit. So if the wire truly is
Cat3, you can use 10Mbit switches and be in good shape.
Now, how do you know if the wiring is truly C
Hello!
I am in the middle of having a vanity toll-free DID set up. It's been 13
days now (9 business days). This is the first time I'm doing this, and
I'm not sure of the process. There has been a very weird progression of
changes on my number, from fast-busy, to a message saying that I'm ca
Hello!
I wanted to make sure that, in addition to my complaints, I make it very
clear: Digium's support is excellent. The jury is still out on the
usefulness of the TDM products. However, Digium has worked very hard to
make sure that this issue is resolved. I actually got an e-mail from
so
[EMAIL PROTECTED] wrote on 03/22/2005 03:56:22 PM:
> On March 22, 2005 03:08 pm, [EMAIL PROTECTED] wrote:
> > The phone in question is what I would consider to be a good-quality GE
> > two-line cordless telephone. Digium's guess is that it is "putting
power
> > on the telephone line and the card
Hello!
I just wanted to tell everyone that I have successfully used a TDM400 with
an IBM Netfinity 5600 server. I used PCI Slot 3 (the first hot-swap PCI
slot). I had a ServeRAID 3L controller in slot 1as well, which managed
the server's array. Other than that, there was nothing extra insta
Hello!
Attached to the bottom of this e-mail is an edited version of an e-mail I
originally wrote to Digium tech support regarding Ouch and Power alarm
errors I have been receiving on my TDM400. It contains a great deal of
detail regarding my setup. In the end, I have found that one of the 5
Hello!
In spite of a number of complaints, I have tried to use a TDM400 on a Via
EPIA-MII motherboard with a 1.2GHz C3 CPU. cpuinfo and interrupts are
included at the end of this e-mail. I have had no problems with it so far
that I can attribute to the computer. I have, though, had continual
Hello!
I have a cabinet full of Wilcom Enhanced Line Powered Amplifiers with
Manual Balance, model ELPA-421V. I *believe* these were used for a bank
of analog modems back in the mid-90's. They were removed from a suite
when the old company moved out. Here's a URL:
http://www.wilcominc.com/e
asterisk-users@lists.digium.com wrote on 02/03/2005 02:20:57 PM:
> [EMAIL PROTECTED] wrote:
> > Also, we're currently looking into toll-free service, but the
alternatives
> > seem to be much the same. At least nobody is telling us if there is a
way
> > to lock in a certain number even if we c
Hello!
We are open to the possibility of changing our business telephone number
shortly. This will most likely be necessary due to a physical move,
changing providers and a few other reasons. However, we woud like this to
be the *last* time we need to do this. Ever. No matter what. Is that
[EMAIL PROTECTED] wrote on 10/07/2004
03:02:41 PM:
> I have an opportunity to pick up a couple of NetFinity 5500's 4 way
Xeon
> 550's w/ 2 gig RAM for very little $$$
>
> I have seen this:
>
> http://www.mail-archive.com/asterisk-dev@lists.digium.com/msg00719.html
>
> In it, there is a passing
Hello!
I've been playing with two pieces of
hardware: a X100P and a TDM400P with an FXO and two FXS modules.
I had been using just the TDM card; however, the TDM FXO module
seems to hear things and "answer" the telephone for no reason,
and I wanted to compare the results with an X100P card.
I
[EMAIL PROTECTED] wrote on 05/28/2004
03:51:02 PM:
> Dear users:
>
> I have bought TDM04B card and it works in PCI 2.2 ver. slot.
> How can I check if specific mother board support PCI 2.2 ver.
> I do not have any documentation for that motherboard.
The easiest way is to look at the chipset. A
[EMAIL PROTECTED] wrote on 03/20/2004 02:58:21 AM:
> You give too much credit to people, indeed. I cannot say about this
list,
> but most lists I use have high corporate populations, where the users
> *have* to use mailers like Outlook or (cringe) Notes. For mailing list
> admins to expect u
[EMAIL PROTECTED] wrote on 01/26/2004 01:12:09 AM:
> You're right, Jeremy. I made up the whole thing. I went out of my way
to
> concoct a story about how I wanted to do business with you, but was
unable
> to figure out how on your website, so I called and left a message and
didn't
> get a re
[EMAIL PROTECTED] wrote on 07/31/2003 12:52:10 PM:
> www.nufone.net is entirely Asterisk/IAX.
You know, I've called them several times and left my telephone number to
call back. I've never heard from them.
You know, many people on the list raved about them. But for a company with
a complet
Hello!
Well, so much for mailing me off-list: not a single person did! In other
words, you've already seen the results of my request:
The options are:
Nufone.net
Cost: 2.9 cents/min for both outgoing long distance and incoming 800
calls. Service is pre-paid.
Advantages: Extremel
[EMAIL PROTECTED] wrote on 07/18/2003 06:11:16 PM:
> On Fri, 2003-07-18 at 17:05, CTI wrote:
> > Does anybody developed Predictive Dialer using Asterisk/Digium PBX?
>
> There has been talk about how to do this, but I don't remember anyone
> announcing it as either done, or open sourced.
Can w
[EMAIL PROTECTED] wrote on 07/18/2003 12:03:06 PM:
> I recommend NuFone. They are completely Asterisk-based, and obviously
> integrate with Asterisk perfectly.
>
> Rates are good, support is excellent, and you'd be indirectly supporting
> Asterisk.
>
> www.nufone.net
I tell you what, if Wade
Hello!
I would like to get connected with a VoIP provider for home. At some
point, I'm sure I will be connecting to it via an Asterisk box, but for
now, I will be using whatever hardware they provide.
What recomendations do you in the Asterisk community have for a reliable
VoIP service that
[EMAIL PROTECTED] wrote on 07/14/2003 12:37:33 PM:
> My fantasy machine for this purpose would be along the lines of a
> mini-itx system with external power supply, dual Ethernet interfaces
> on board, and one PCI slot available. If it had one real serial
> port on it, that would be great too
[EMAIL PROTECTED] wrote on 06/08/2003 11:07:40 AM:
> Thanks Steve..
>
> Regading the HOW SMALL I think the question is HOW MUCH $ , I`m thinking
of a
> more-less $200 or $300 usd BOX working as a Broadband router , 2 NIC`s ,
> asterisk on it, With Optional a WIC like slot to put fxo's or Fxs
[EMAIL PROTECTED] wrote on 03/03/2003 09:02:52 AM:
> I don't think I'd be stepping on toes if I told you guys we're paying
about
> $35/month per analog line with callerid, line hunting, call waiting,
etc, etc
> (basically the works minus voice mail) through AT&T
>
> That's Canadian so you'll ha
[EMAIL PROTECTED] wrote on 03/03/2003 12:25:11 AM:
> On Sun, 2003-03-02 at 05:40, [EMAIL PROTECTED] wrote:
> > I've heard others say that PRI becomes cost
> > effective in the 8 line range, but the cost for the office with 14
lines
> > ($25/line * 14 lines) is only $350. It would take 22 lines
Hello!
Several of my customers would like to add a backup to their Internet
connection. ISDN is a good solution: decently fast for a dial-up-type
connection, yet still faily affordable. While I was at it, I decided to
look at a couple of more creative telephone service options to possibly
i
[EMAIL PROTECTED] wrote on 02/26/2003 01:31:33 AM:
> This
> certainly isn't as cheap as the $40 Actiontec interface, but it will be
> fully supported by Digium, and of course proceeds will support further
> development of Asterisk. The station card is currently in production
and
> is expected in
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