,
and this problem has existed since FC1, Mandrake10, and probably others.
Wouldn't you think that would have been corrected by now?
John Novack
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you already know the answer.
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don't remain off for a long period of time. All
the addresses behind the router that would interact with Asterisk are
fixed, and the necessary ports opened on the router for IAX to work.
Not all ISP's are so kind, I am led to believe, but have no firsthand
knowledge.
Perhaps some more information can be
loads about the flexibel (sp?)
rate not fully tested.
The Wiki mentions this, but again a solution is not provided for the novice.
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Neil A. Hillard wrote:
<>John,
In message [EMAIL PROTECTED], John Novack
[EMAIL PROTECTED] writes
Perhaps the instructions have improved, but I discovered that some instructions were incomplete. Perhaps not to a Linux expert, but to the "normal" reader, information
When issuing a "stop now" to Asterisk, the message "Yuck! Error in
buffer handling...: Success" is returned.
No complaints when Asterisk is started, and everything seems OK while
running
.
Google provides no help
RH9, CVS-HEAD of 2/24/2005
Any clue as t
config to the list within the past 10 days or
so, so a search through the recent archives should bring a working result.
The key here is the MAIN Vonage number is locked to the ATA, but a
softphone add on account from Vonage should work.
John Novack
If you want to use Asterisk, you will need
line POTS phone, with or without answering
machine, Panasonic is the one to buy.
John Novack
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their replies.
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automatically.
Have to wonder why this known ( for more than a year ) problem has yet
to be fixed.
John Novack
Roger Hanson wrote:
See
subject:
Does [EMAIL PROTECTED] support Dual-Processor installations? I didn't see
anything on the sourceforge page clarifying that. I suppose they could
leave
Francesco Peeters wrote:
On Fri, April 1, 2005 22:04, Tim Bass said:
Dear All,
Thanks for your help :). As I mentioned earlier, I don't have any issues
with email, except that I thought most advanced communities had moved past
SNIPPED OFFENSIVE CONTENT
Jean-Michel Hiver wrote:
Kerry Garrison wrote:
The book bills itself as a beginner's guide to Asterisk and Voice
over IP (VoIP). Even with over 270 pages, it isn't possible to go
through every single feature that Asterisk has to offer but the book
does give enough information to get you started
Brian Capouch wrote:
James Gardiner wrote:
Asterisk 2.0 on Windows.. This is all very much a bit of a Joke, but
one
does beg to ask.
When will version 2.0 be released???
2.0 is just now really being talked about in earnest.
I think a better question would be when 1.2 is going to be out.
An even
One would hope so, but one
of the fist posts I see is someone ranting on about how if you haven't
read x or y you don't deserve an answer.
It is that kind of a social misfit that should not be welcome anywhere,
but seems to have too loud a voice here.
JN
Ty Carter wrote:
Thank
you for your
Well, you COULD use your
delete key.
You DO have one, don't you?
And you complain of others posting stupidity
JN
C F wrote:
What can be done to this shmuck?
Everytime I post anything to the list I get one of these. I'm sure
I'll get one for posting this one as well.
--
the last 24 hours.
No clue if it is for this or other problems
John Novack
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- Original Message - From: Rich Adamson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
access to a line.
John Novack
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irst article, and half the second, much less flexible,
position..
The really curious thing on this list is every so often, if I choose to
reply, the poster AND the list appear, but mostly just the list, as if
the poster had some control as well.
on the board it is revision H.
Curious.
FWIW, I use the clone card for FXO, and have no problems with any
crackle. The echo problem resolves itself after a few seconds of
training but I am using it on a VERY short loop to connect to an
electromechanical step by step switch
John Novack
for now.
John Novack
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Why would someone choose
these over other boxes, such as the Sipura 2000 and 3000?
Also, anyone have experience with the TDM400 FXS module in a ground
start configuration?
Digium has said , from two different support folks that they do not and
do work ground start.
John Novack
Colin
current
shortcomings ?
>From reading through the archives, it seems there is currently no way
to reset to factory default, no written MAC address on an individual
box, and some other instabilities requiring frequent resets.
Digium support also seems rather slow, to be kind, via E-mail
John Nov
until all are busy, then you
either need to return busy or perhaps go to a VM box?
Sorry I don't have an answer for you.
Are there different types of queues or hunt groups that you can define?
John Novack
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of, ( junk fax reception in the US is covered under some earlier
telecom law, and senders can be subject to fines if they can be found )
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s what they try to direct you to, as they charge something for
that one.
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the
set from the TR of that line.
Many 1A2 multi line phones could be connected in parallel, in an office.
The last 5 pair on this phone were probably used for outboard
speakerphone, and would not be used or connected to other sets.
Elementary Telephony.
John Novack
via the internet, and find the support for such basic
telephony somewhat lacking.
Interestingly enough, it seems the chipset used on the TDM 400 module
supports a ground start configuration, but Digium chose not to make that
available.
John Novack
machine.
JMO
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.
The only way to handle is to dedicate extensions and ports for each
company, a real waste of resources.
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Brian Roy wrote:
On Tue, 15 Feb 2005 17:12:01 -0500, John Novack
[EMAIL PROTECTED] wrote:
The real problem with Panasonic and anyone's voice mail is tenant
sharing. Calls to VM or returning to VM don't contain the trunk number
information.
The only way to handle is to dedicate
? I see that the 1750
is listed
on the Wiki. How have others powered the TDM400P in a Dell 1750?
Have you no way to use one of those power "splitters"?
One female to two male cables that can be found almost everywhere.
Unplug a pwer connector to a HD or CD and insert.
J
ound start is an important feature for those of us who are using
Asterisk as an interface to our electromechanical switches,
interconnecting in a private collectors network.
John Novack
Michael Welter wrote:
Chris
Blake wrote:
Greetings *`s,
I have a Digium TDM01B card which I want
is a rehash of what is available on
line free for the taking.
Just my opinion, of course.
John Novack
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And yet another "helpful"
comment to clog up the list.
Some people use HTML
Some people top post
Some people don't read too well
Some people aren't as skilled as others in searching.
GET OVER IT
JMO
John Novack
Jens Vagelpohl wrote:
On Feb 21, 2005, at 23:36, [EMAIL PROTEC
RIGHT ON!
Too bad you also didn't post in HTML as well
Perhaps this list needs to be split?
One for the folks who simply want to get it working, and another for
the self appointed list police who want to be rude and nasty and are
only interested in feeding their own egos, all the while
is good connecting the two?
Are you sure the FXO card is good?
Output of the ATA should appear to Asterisk as a PSTN line.
Perhaps your problem is really simple?
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Christopher wrote:
I just would like to not have that damn status light flashing all the
time. It hard to explain to people who walk in the server room :)
A small piece of electrical tape works wonders!
Also works well on "check engine" lights!
J
is poorly supported.
John Novack
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that are fed up with their postings should
just forward those back to their email address. Maybe they would get the hint
and some manners.
Nah!
They were never taught any social skills, and it's too late now.
John Novack
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Anyone using this Sip phone with Asterisk?
If you have had success getting the message waiting indication to work,
please contact me off list.
TIA
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The Phone Guys wrote:
So you want it 100% perfect and you want it for peanuts.
OF COURSE!
They all certainly imply and promise that.
Would anyone subscribe if they said we have a second rate service ?
Makes you wonder how many *really* reliable VoIP providers there are
out there?
Who would
choice.
Personally I don't think any SIP phone needs to cost much over 100
bucks. and could still be usable and rugged.
JMO
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hat number.
There was some mention of this on the list a week or two ago, and soon
the "list police" will yell at you for not searching first through the
list archives and the wiki, usually with less than satisfactory results.
John Novack
Frank Abernathy wrote:
I a
Sangoma gives EXCELLENT technical support.
I would suggest you try there first.
The few problems I have had with installation were addressed promptly
and when driver fixes proved necessary, corrected in short order.
Also the cards have a 5 year warranty!
John Novack
Nhadie Ramos wrote:
Hi
files, install
CentOS 4 or 5 and use the X100 as a paperweight. I suppose it is OK for
a timing source, but not much more.
My opinions, worth what you paid for them.
John Novack
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as well with a 5 year warranty.
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Stephen Bosch wrote:
Well, this is approaching the absurd.
Do you know how many Meridian systems have radios plugged into them for
on-hold background sound? Nobody pays royalties on those.
IF they are discovered by ASCAP and receive a letter demanding payment
they will. Not absurd at
Stephen Bosch wrote:
John Novack wrote:
Stephen Bosch wrote:
Well, this is approaching the absurd.
Do you know how many Meridian systems have radios plugged into them for
on-hold background sound? Nobody pays royalties on those.
IF they are discovered by ASCAP
.
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Paul wrote:
The thread is about music on hold. Things such as playing local radio
stations in a waiting room are not related. I don't think there is anything
illegal about using normal over the air radio and TV for such purposes as
long as it stays in the local market area.
It is ALL
IS the standard, and until VOIP reaches that level
of reliability and quality, there will be many business failures
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be alright
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in ALL modern machines.
If they can't make it work ( never seen that ) they will refund.
If you have problems, and you give them SSH, they will fix it.
John Novack
Steve Totaro wrote:
That is why I suggested Sangoma. Ask them if you can return it if it
does not fix your problem
Anthony Kepler wrote:
Thats AMAZING! This google you have shown me is truly a modern marvel
of the interwebs.
You know what would be EVEN BETTER though?
If idiots (such as you and I) would find something better to do with our
time than mock others on mailing lists in a pitiful attempt to
, and a ground on the
subscriber end will introduce hum, or if low enough resistance will keep
the line in an off hook condition.
Swap the Adtran cards if only on some circuits, and make sure the 750 is
well grounded.
John Novack
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that would do what you want.
ValCom and Viking come to mind.
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Been covered Ad Nauseum on the list.
Asterisk does NOT detect dialtone
w, or a series of w's befor dialing begins will help, EXCEPT when doing
pulse dialing.
w does NOT work with pulse dialing
No one seems to think this is a problem, so it doesn't get addressed.
John Novack
Joseph Tanner
that way, and have for many a year
Most users expect transfer to work that way.
I would consider that a defect or bug, not a new feature request.
John Novack
Ira wrote:
At 12:57 AM 02/12/2006, you wrote:
Why don't you think it is correct behaviour? The purpose of attended
transfer is that you
if the callerid number field
doesn't contain a string of digits, or, even better, a string of digits
of a length that can be specified, all without breaking the callerid
name..Replacing any callerID name delivered with privacymanager seems
unnecessary.
John Novack
!.
Seems the designers thought everyone in the world now used DTMF.
Perhaps someone else has a method that will work. w will not.
John Novack
Balint Kovacs wrote:
Hi Grigoriy,
Thanks for the reply. I have tried to implement this dial pattern by
dialing from 8w10 to 8ww10, 8p10 (which
Most/all assemblers have a better and more consistent parser though.
The parser for the extensions dialplan is just short of insane
John Novack
Douglas Garstang wrote:
Yes, programming the dialplan is akin to programming assembler.
-Original Message-
From: Anthony Rodgers [mailto
. Either
Asterisk needs to change to move into this market, or another product will
JMO
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, probably
more, require 7 digit dialing to another state for one NPA, and 11 digit
for it's overlay.
Mobile providers in the US usually require only 10 digits, and fill in
the 1 within the phone.
The only rule is there are no rules. The result of local/state rule.
John Novack
sdgesa gaeharth wrote
You should also know that this ONLY works with DTMF on analog lines. If
one happens to have to use pulse dial on a POTS line, there is no way to
delay dialing, and Asterisk STILL will not wait for dialtone, since no
one who is able to fix it seems interested.
John Novack
sdgesa gaeharth
Tim Nelson wrote:
Thank you all for the suggestions. I'm looking into getting groundstart lines
for that installation as suggested earlier.
Make sure your interface supports GS
The Sangoma and TDM cards do
I assume you are using one of these as you mention Zaptel.
John Novack
Also, I'll
shadowym wrote:
Ok so I'm not going crazy then.
The jury is still out on that issue!
John Novack
I filed a bug report.
http://bugs.digium.com/view.php?id=12093
-Original Message-
From: Trevor Peirce [mailto:[EMAIL PROTECTED]
Sent: Wednesday, February 27, 2008 6:46 PM
Not a good application for Asterisk.
Plenty of good used hybrid systems around for cheap
With the failing business and economic decline in the US there is a glut
on the market
Start with eBay
Toshiba, Avaya, Lucent, NEC or Nitsuko
Panasonic is probably the best choice of all , since they have
that was installed 20 years ago, and NEVER needed
any attention until it finally failed.
John Novack
Andrew Ladanowski wrote:
http://www.3cx.com has a free system for non profits that is easy to
use.
It is windows based and runs great on XP or a server.
Grandstream - HT386 is easy and cheap. It it less
.
John Novack
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to version, sometimes
completely out of the realm of expressed policy, that may not break the
average users application but bites our Tandem application.
Just my opinion, worth what you paid for it!
John Novack
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An NEC DSX with CF voicemail and e-mail integration wholesales for well
under 3K, double that and add cabling . . .
Well, you get the idea.
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that
were in use before semiconductors were even in a engineer's wet dream.
Clearly the Sangoma is a better card in many ways, but your needs would
require you expand the A200 to do what you want.
That ultimately may be the best solution.
John Novack
John Novack
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Sometimes it's the really simple things that grab us!
John Novack
Thomas Klettke wrote:
On Sat, 2008-03-22 at 12:48 -0400, John Novack wrote:
Assuming you have also checked the obvious possible defects regarding
cords from the XO device to the Digium card, what happens if you
BUT - does it cure baldness and impotence?
Shouldn't anyone who posts stuff like this be banned? After flogging
tarring and feathering, of course!
Peg Leg O'Brien
BerkHolz, Steven wrote:
Asterisk work does not pay all of my bills, so I have joined up with a
company that has a very good
numbers. Many telcos provide a group of lines or
trunks under one number with multiple appearances in a hunt group, often
tarriffed at a different rate.
Probably more than anyone wanted to know about Direct Inward Dialing
John Novack
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WOW! Is this LONG overdue.
Why this wasn't done initially is beyond me
It has caused so many troubles and questions and posts from folks who
expected Asterisk to at least have a feature that has been in a dial up
modem for 10+ years.
Great job!
Many thanks
John Novack
Steve Davies wrote
Well, it SHOULD have been obvious that IF I had degree of skill I would
have years ago.
Those kind of comments serve no purpose other than to anger and to boost
the already inflated ego of those who make the comments!
This is a USERS list after all!
John Novack
Tilghman Lesher wrote
a small inductive kick when it opens ( goes
on hook )
John Novack
Marlon Dutra wrote:
Hi guys,
I've been experiencing a very strange issue with my Digium Card TDM400
as of this week. It has two FXS and two FXO.
The FXO modules (both of them) never goes on-hook after hanging up in
Asterisk
hardware platform
as well leaves much to be desired in reliability.
John Novack
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, and has been true for many years, even in
electro-mechanical systems.
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cell phone is then answered by cell phone voicemail
instead of asterisk voicemail.
Any ideas how to go about this?
Thanks!
Steve
Zap channels are considered answered once dialing is complete, so your
10 second time fails
No answer supervision on Zap
John Novack
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, and no
one who has the skills ( I certainly don't ) is interested enough to
correct it.
Analog PSTN interfaces will be around for a long time. It deserves some
attention.
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The list police are out in force today!
More archive space is used up in these kinds of complaints than the OP.
Let's move on.
Peg Leg O'Brien
Erik Anderson wrote:
On 4/24/07, Astawerks [EMAIL PROTECTED] wrote:
No virus found in this outgoing message.
Checked by AVG Free Edition.
I bought one of the Plantronics 120 or 130, not sure which, and the
acoustic conduction within the device causes such an echo that it is
unusable with my 842. Even the speakerphone on the 842 is better, and
that isn't saying much.
John Novack
JR Richardson wrote:
I have 4-5 different
Just because you are paranoid doesn't mean they aren't out to get you!
Frederico Madeira wrote:
Hi guys,
During a time, i'm looking for a softphone that work fine in linux,
present good features, good audio quality and good interface.
Recently i found gizmo and i'm testing it with my
A common attitude in the development community.
Keep adding more bells and whistles, it's more fun and interesting.
Don't bother to fix the many existing problems. That is boring
Peg Leg O'Brien
Steve Totaro wrote:
Rather hasty I think. I think whatever version 1.2.X winds up on should be the
How many times will this get posted??
Salah Eddine ELMRABET wrote:
Hi All,
As Net2Phone don't permit more than one session per account, I
configured about 10 sip trunks and configure multiple trunk routing
but once the first trunk is used I cannot make additional calls, I
also cofigure my
Sangoma offers EXCELLENT Technical support
Why not try them first?
They have never failed me yet, even for our peculiar requirements
Email address on their website
John Novack
Sanjay Rajdev wrote:
I have a 2 sangoma cards that need to be configured on a same server, one is a
T1 and another
bigots that
insist on using their favorite, and end up with problems.
There are those who Linux is simply a means to an end, and not a religion!
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is
ready, and it may miss a digit, causing misdials.
Curious that cheap modems years ago learned to listen for dial tone,
but the Zaptel driver doesn't, and of course this is considered a
feature request rather than a bug, and no one seems to want to fix it.
John Novack
such complaints
John Novack
Rob Schall wrote:
But if this was the case, then why would the message playback (from the
provider) read back the digits from the start. I mean, I dialed
630-XXX-XXX and it reads back 312 (my local area code)-630-XXX-X
I would think if it wasn't waiting, then it would do like
Not option w
add a w in the DIAL STRING
Guess you didn't search the archives?
Sometime in 1.2 this feature was fixed to work with pulse dial as well.
example:
exten = s,1,Dial(ZAP/g4/w(${ARG1:3:4}),360,Tt)
John Novack
Rob Schall wrote:
Also, in the dial command the w says its
Well, the claim of 1 members MIGHT have something to do with it!
E-mail delivery is notoriously erratic as well.
If you think this is slow, try some on Yahoogroups!
John Novack
David Boyd wrote:
Can anyone enlighten me as to why it takes 40 minutes or more for a
posting to the list
Hearing the noise might be of some help.
It seems as if it might be some sort of digital noise.
I have an A200 working, but no expansion board with no such noise.
Also I have NEVER had to run the gain as high as some others suggest on
the FXO ports.
John Novack
François Delawarde wrote:
Hi
transmitters could certainly
generate a similar noise.
I am assuming that several different sets have been tried with similar
results.
My experience with the Sangoma A200, and in fact Digium 400 and Adtran
channel bank FXS circuits all are very quiet, so this is a bit of a puzzle.
John Novack
do you want for 20 bucks anyway??
And when will Digium offer affordable one-port cards again?
Probably never.
John Novack
-HJC
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in progress??
John Novack
- Original Message -
From: Adrian A [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, June 6, 2007 2:50:00 PM (GMT-0600) America/Chicago
Subject: [asterisk-users] Voicemail marking
Stephen Bosch wrote:
John Novack wrote:
I don't know what configuration changes I would make to solve this. My gut
feeling is that it's an electrical problem somewhere within the system, but
perhaps I'm reaching for that too soon; in any case, I wouldn't know where to
start even if we
provided by Sangoma.
John Novack
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rolled into their standard set, as
David's English skills aren't up to his excellent support skills!
Don't even ask why about pulse dial - too long a story!
John Novack
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