Michiel van Baak wrote:
On 01:30, Wed 13 Aug 08, Ronald Wiplinger wrote:
I had installed in the office an Asterisk server, but the company is
gone and I could keep the server.
However, for my family with three members and two phone lines this
server is overkill. I am looking for a compact
this would
permit internal calls to resume, as long as there are no attempts to
dial external numbers.
dnsmasq is just the creature for the job. Very flexible and easy to
configure.
http://www.thekelleys.org.uk/dnsmasq/doc.html
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a raid 10 array in the 1950, but you'll have to pay for the
hyper-expensive 2.5 drives.
Darrick
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mine
within a few hours.
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used. The D-Link's have been very reliable.
Darrick
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chip that is
fully i586 and partially i686 compatible. If you have a distribution
that is compiled with i586 optimizations, you won't have problems.
Darrick
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David Nedved wrote:
--- Darrick Hartman (lists) [EMAIL PROTECTED] wrote:
Do yourself a favor and upgrade a Asterisk 1.4 which has a proper
implementation of DTMF. It's likely your SIP provider upgraded to
something which does not recognize the DTMF tones from Asterisk 1.2.
I've upgraded
to an extension in the meantime, not an elegant solution.
Do yourself a favor and upgrade a Asterisk 1.4 which has a proper
implementation of DTMF. It's likely your SIP provider upgraded to
something which does not recognize the DTMF tones from Asterisk 1.2.
Darrick
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have no incentive to offer
such a hand-off because they wouldn't make any money on the calls after
they are handed over to the voip system.
Darrick
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and reliably across a wider
range of Linux distributions.
Great! What will it take to get the g729 codec module compiled against
uClibc?
Thanks,
Darrick
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Doug Lytle wrote:
Evan Ruff wrote:
Since when is the users list a transport for calendar scheduling?
Since when are humans infallible? Randy made a mistake. He apologized
for it. Let's move on...
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or recommend something post with your real identity. That way people
can make a reasonable conclusion on your advice.
Who knows...perhaps the Shadow knows [1].
[1]: http://www.mysterynet.com/shadow/
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a /third/ level of configuration in
Wanpipe.
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any sense as a dependency.
Darrick
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with parking calls.
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1.4 that all are behaving just
fine. My major motivation for moving to 1.4 was DTMF.
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bilal ghayyad wrote:
Hi;
Via OpenVPN or port forwarding is known for me, but
via SSH is new for me, how I can do it and what is the
difference by SSH and OpenVPN?
SSH uses tcp. Openvpn, by default uses udp.
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work with TCP
traffic. IAX2 uses UDP packets, so I don't think that'll work. You
might try setting up a VPN or something along those lines. (Also, IAX2
defaults to port 4569, not port 5060.)
OpenVPN works great for this.
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for the ZAPMODS variable. You should
have that variable set to:
ZAPMODS=wctdm
Beyond that, as long as Asterisk is not running, issuing service zaptel
stop should remove all zaptel related modules.
Darrick
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Vincent wrote:
On Wed, 09 Jan 2008 06:01:32 -0600, Darrick Hartman (lists)
[EMAIL PROTECTED] wrote:
But look in your /etc/rc.conf file for the ZAPMODS variable. You should
have that variable set to:
ZAPMODS=wctdm
Yes indeed:
#ZAPMODS=wctdm
Should I add this module here
the way into the handset. I've seen some Polycom
handsets that look like they are plugged in, but in reality, the end of
the cord that plugs into the handset needs to go in farther.
Darrick
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has been correctly registered!
That card was picked up by hisax?
I doubt it. My guess is he never rmmod'd zaptel or rebooted the box
before trying to modprobe opvxa1200. He was probably running the
original zaptel module which didn't know about the new hardware.
Darrick
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it operates with Outlook 2007,
I'd say it's not usable. I have not tried with Outlook 2003, but since
I need a working solution for both Outlook 2003 and 2007.
Darrick
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for together on the free one and
post here in a bit.
r
Can you just install limesurvey on a server some place? It would allow
you to do however many future surveys you want to do.
Darrick
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,
There must be something in the water (or wine) in France. Nothing on
the limesurvey site requires you to register for anything. It is very
current and updated about once a month (far from abandoned). Perhaps
someone else made a different suggestions.
Darrick
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.
Good luck!
Darrick
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. What format are you currently using?
Darrick
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case, I applaud the author of Askozia for his efforts. Taking on
a project of this size takes a considerable effort. In many cases it's
a volunteer effort.
Regards,
Darrick
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(hosted off the sourceforge project page).
Note that the net5501 platform was added recently in trunk. None of
the images have been released yet, but several of us have been running
releases from SVN for several months.
Darrick
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) or an A200 can support up to 24
FXO ports. (6 spaces)
I can't comment on how good they are, I've only got TDM400Ps myself.
Rhino also makes some very nice cards and have a good support staff.
The Rhino cards are also made in the US.
Darrick
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I'd say this sort of thing belongs only on the biz list, but
this sort of issue may affect so many people it's worth noting here (but
not dragging out with hundreds of me toos).
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,
Look at his path. He's going from a PSTN phone to a g729 gateway. As
long as the gateway is there, Asterisk doesn't really know about the
PSTN phone. Therefore, yes, this should equate to pass through.
Darrick
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Peder @ NetworkOblivion wrote:
Is there a way to decrease the volume on the native files version of MOH
in 1.4? I've had several people complain that it is too loud.
run the files through sox
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use them in a new install.
Darrick
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Dermot Bradley wrote:
Darrick Hartman wrote:
Just because someone is using an old kernel or doesn't know what they
are doing doesn't mean the hardware is bad. I've had very good
success
with dozens of different VIA boards (from the original mini-itx board
up
to current C7 models
(or not as the case is!)
Gordon,
I don't think you're doing anything wrong. My experience with these
little creatures is similar to yours. They perform well under load and
just plain work.
Darrick
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and compact.
Not sure what else you need to know.
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). If you are doing a ton of transcoding or recording calls,
your results may be different.
Darrick
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does 60/6 rounding. You only pay for the first full minute, then
fractionally there after.
I've been using them for over 2 years with only a few issues that were
quickly resolved.
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of a transcoding issue than anything else.
Darrick
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sure your sip.conf entry with
the dtmf=rfc2833 is being used.
I'll chime in since nobody has yet corrected this... it's
dtmfmode=rfc2833 not dtmf=rfc2833
Yeah that's what I meant. Total lack of caffeine this morning.
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personal box. My sip
provider went un-reachable (Teliax requires the use of hostnames). When
that happened, I couldn't even call my local phone extensions.
Everything SIP was locked hard until it finally timed out.
Darrick
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or update options visit:
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Small Business IT Specialists
Office: 920.547.4535
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all excited about it only
to be terribly disappointed when I unpacked it.
Actually the original poster did say this was a Windows program. I
believe the exact wording was It runs on any modern flavor of Windows.
Note that Mats Karlsson was NOT the original poster.
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(properly) and become part of the code that
will benefit everyone. His request is not a blind add this feature for
me because I'm a leach. Anyone using Asterisk in a business
environment should care about this and want it resolved.
Darrick
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http
.
Openvpn can be configured to establish either a TCP or UDP connection
(with UDP recommended and shown in most examples).
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at Astlinux. We
will be releasing 0.4.5 in the next week or so with some major
upgrades/improvements. For now, you can grab the release candidate
images from here: http://www.djhsolutions.com/astlinux
Darrick
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of read-only mirrors of the
site available as alternate access.
Thanks for using voip-info.org!
[EMAIL PROTECTED]
end quote--
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explained here:
http://www.opengroup.org/onlinepubs/009695399/basedefs/xbd_chap08.html
After making the change, the easiest thing to do is reboot. That will
re-create the /etc/TZ file (well actually /tmp/etc/TZ) on your system.
Darrick
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Whoops! This message was intended for the Astlinux mailing list.
Darrick Hartman wrote:
Until the new timezone data files are updated, you can correct the
time on your system by using the following:
Using Central Time Zone (US) as an example, in rc.conf replace
TZ=CST6CDT
with
TZ=CST+6CDT
one system to another. Most distros had
their update files in place a while back.
Darrick
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Dinesh Nair wrote:
On 02/25/07 06:26 Darrick Hartman said the following:
Kristian is working with Sangoma to get wanpipe supported once again
in Asterisk.
is there a reason why wanpipe stopped working with asterisk ?
I meant with AstLinux. Sorry for any confusion.
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once again in
Asterisk. There have been some recent (last few days) changes in svn
that indicate we are very close to having this working.
Darrick
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protocol for teliax? I
consistently had poor quality with iax2 via teliax. Switched to SIP
about a year ago and haven't had an issue since.
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with the L for Linux.
Newegg has them for $57 after MIR.
Darrick
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that.
Darrick
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from a BSD background, you might find your way around Slack
easier than other distributions.
If you want something dedicated to Asterisk, you might want to give
Astlinux a look ( http://www.astlinux.org ).
Darrick
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with Windows? If
reboots are requires, then there is a memory leak or other problem that
needs to be addressed. Rebooting is not the solution to other problems.
Darrick
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for new business. In other cases, no they are not able
to take that risk so you plan around it.
I guess what some people are trying to tell you is there is another view
other than your own.
Darrick
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and understand the risks and make
appropriate business decisions based on your business.
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versions but only SVN trunk (sometime to be 1.4)
[1]: http://ftp.digium.com/pub/telephony/sounds/releases/
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distros so you might want to google for the
fixes/issues before pulling your hair out.
Darrick
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and we could
probably work something out.
Darrick
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Please let me know the problem ASAP. Looking forward to your response.
Here's an idea! Contact Teliax support. I have had no problem making
or receiving calls.
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selected, or when
jumping to the
'o' or 'a' extension.
Options:
f - Allow the caller to enter the first name of a user in the directory
instead of using the last name.
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behind on the phone. The board seems to handle the
load quite well. Additionally openvpn is actively used as this box also
serves as the firewall.
Darrick
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will be welcomed.
For information about AstLinux, go to http://www.astlinux.org Note that
there have been some pretty substantial changes in the past month and
some of the documentation hasn't quite caught up to the 0.4.0 image.
Darrick
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Andrew Kohlsmith wrote:
On Thursday 18 May 2006 08:07, Darrick Hartman wrote:
Don't attempt to use a Digium or other FXO and FXS card on a soekris
board, especially if you plan on using a full-featured distribution like
CentOS (which is what AAH uses). See more below...
I read below
work. For
the advertised feature set, these looked impressive.
Darrick
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Seth Remington wrote:
Anybody have any recommendations? IAX service preferred.
Teliax has been good to me.
Darrick
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for this? :-)
Then, in extensions.conf, set a hint for the _watched_ extension like this:
exten = 2348,hint,SIP/2348
Very good explanation. Additionally, (at least on the Polycom 600's)
you need to reboot your phone for this to take effect.
Darrick
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to them cleared it up. Email
support is spotty. Sometimes great, sometimes unanswered for a while.
Calling them has been the best way to get support the few times I've
needed it.
Darrick
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there.
Darrick
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is not an speaker phone. It only has
a listen-only hands free setup.
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choice, but you will need to provide power
to the PCI card to handle the FXS ports, plus the digium TDM cards don't
fit in the standard case.
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. Nothing wrong on this side. There does appear to be something
wrong that someone at Digium should look into deeper.
Darrick
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period. You're only gonna waste time.
If you really insist on trying, buy a second DID and register that one
with g711 only.
Darrick
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), but they have their limitations.
Darrick
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the dial plan alot
easier to read. Why not make things easy when possible, rather than
forcing a newer user to have to jump through all sorts of hoops to
implement a simple feature like this.
Darrick
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missed here? (I.e. please help!)
2) Is there a document I should be working off of? Google doesn't seem
to think so...
http://iaxmodem.sourceforge.net/
Look there. It might help you ;)
The other suggestion would be to ask on the iaxmodem mailing list. Join
us over there.
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this:
exten = s,1,Wait(1) ;sometimes you need to wait to get callerid
exten = s,2,Answer()
exten = fax,1,Dial(IAX2/200)
You can't tell if it's a fax until it is answered. Read up on the use
of the 's' extension.
Darrick
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to work. The current silence
threshold is set at the default 128. The FXO module is configured with
Kewl Start signalling. Would I get better results (or any different
results) if I switched to Loop Start?
Any other ideas?
Thanks,
Darrick
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http
the gain
on the file before sending it off?
Could you provide a little more information? Is this incoming VOIP or
incoming via a Zap channel? If it's Zap, what hardware are you using?
Did you try increasing the rxgain?
Darrick
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Andrew Kohlsmith wrote:
On Tuesday 03 January 2006 20:14, Darrick Hartman wrote:
I'm attempting to use an asterisk box with a Digium TDM01B as voicemail
for an existing Meridian/Norstar PBX with an ATA-2 adapter. We're
having problems where hangup is not always (but sometimes) detected.
It's
), is there something I can try to improve the percentage of calls
that are disconnected?
I would use silence detection in voicemail.conf but wanted to know if
there was some other method.
Thanks,
Darrick
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877.901.3113
Leo Ann Boon wrote:
Darrick Hartman wrote:
I'm attempting to use an asterisk box with a Digium TDM01B as
voicemail for an existing Meridian/Norstar PBX with an ATA-2
adapter. We're having problems where hangup is not always (but
sometimes) detected. It's not detected probably 70
are available (as well as
older versions).
Darrick
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a question regarding everybodys experience with
Teliax
or Broadvoice. I setup a Teliax trunk this morning, and had calls
going out
it in about 5 minutes(Had to get more coffee). Has anybody had any
problems
with them, outages, issues with dids etc??
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this up so that sound is recorded?
Thanks,
Darrick
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C F wrote:
What codec are you using?
ulaw
I don't see what the codec would have to do with this though.
On 12/8/05, Darrick Hartman [EMAIL PROTECTED] wrote:
Noah Silverman wrote:
Moj,
It is set as the default. *1
When I dial *1 I actually see user pressed *1 to start recording.
I
]: res_musiconhold.c:488 monmp3thread:
Unable to spawn mp3player
s
Unfortunately, you're the only one who can fix this. Put some mp3 files
in the directory that is mentioned in the error messages.
D
--
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com
would not need to
have a license.
Darrick
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Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com
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to the
appropriate user. I'm guessing that should be able to be done in the
dial plan. Anyone have an example doing this?
Thanks,
Darrick
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Darrick Hartman
DJH Solutions, LLC
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to
record their name instead of a complete busy or unavailable message.
Darrick
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Darrick Hartman
DJH Solutions, LLC
877.901.3113
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that would prevent me from mirroring this as a complete pdf
rather than individual pdfs?
Darrick
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DJH Solutions, LLC
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instructions and some later AMD
instructions, but they do have mmx.
just a question: anyone has never installed g729 codec on VIA
motherboard with C3 processor ?
I'm having problem with IPP libraries, and Intel said that it works only
on Inter processor.
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Darrick Hartman
DJH Solutions
to find something that will work well.
Darrick
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Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com
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