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
you already know the answer.
John Novack
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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.
John Novack
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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
,
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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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
choice.
Personally I don't think any SIP phone needs to cost much over 100
bucks. and could still be usable and rugged.
JMO
John Novack
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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
to get that DB, I don't know,
Short answer is, you can't
AFAIK, it is only available to ILEC and CLEC's
John Novack
but it sure would be nice to integrate something like this
into *, wouldn't it?
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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
John Novack
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http
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 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?
John Novack
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http
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 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
made after 2000 probably is OK
John Novack
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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
? 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
.
The only way to handle is to dedicate extensions and ports for each
company, a real waste of resources.
John Novack
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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
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
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 )
John Novack
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s what they try to direct you to, as they charge something for
that one.
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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