I need one more help can u plz explain what zaptel does it exactly.
This discussion probably needs to move off the dev list. Zaptel is
the set of driver modules that connects hardware from Digium and other
various vendors to Asterisk through chan_zap. This information is
available if you look
of
performance. Which one you purchase depends on what interfaces you
have available in your server.
William Moore
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Has anyone ever had any problem with the TE220B card with it showing up
as four ports instead of two. I RMA'd the first one with the retailer
(Digium tech advice), but I just got another brand new card and it is
coming up as four ports again. The card identifier is showing 0420 when
I do
* I was asking if the endpoint send a call, and it has
a username and password typical to that configured in
SIP.conf file, then should this end point being
registered or not?
If you are only *SENDING* calls to asterisk and not receiving, you do
not need to send a registration. You only need
On 6/26/07, Mike Hammett [EMAIL PROTECTED] wrote:
I am looking for a gateway that has several FXS ports and uses IAX. I have
a need for 16 ports, but will accept 6 or 8 port gateways as well.
Here is an 8 port gateway that should suit your purposes:
On 6/27/07, Ed Nuñez [EMAIL PROTECTED] wrote:
What is a god Windows application to read core dump files?
No. Core files must be examined on the same system that created them.
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On 6/19/07, Carlos Garcia Mujica [EMAIL PROTECTED] wrote:
With what command can I execute chanspy throw Asterisk Console.
THANKS.
You are less likely to be answered if you spam the list.
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On 6/14/07, Matt Scott [EMAIL PROTECTED] wrote:
I purchased FXS modules so that I could terminate the machines or faxes (eg
just like a standard phone) the outgoing/incoming channel will be be
provided by my E1.
I hope I have the right modules for the job?
You do indeed have the right
On 5/25/07, Alex Balashov [EMAIL PROTECTED] wrote:
In reference to an old post from 2002:
http://www.marko.net/asterisk/archives/0203/0103.html
How does one go about doing this?
I think what mark was referring to there is dynamic spans. They
actually work over a standard ethernet network.
On 5/25/07, Matthew J. Roth [EMAIL PROTECTED] wrote:
List users,
This post contains the benchmarks for Asterisk at low call volumes on
similar single and dual-core servers. I'd appreciate it greatly if you
took the time to read and comment on it.
Are you recording memory figures as well and
On 5/24/07, Jeremy Mann [EMAIL PROTECTED] wrote:
Can an asterisk box equipped with a Digium T1 card handle Integrated T1
circuits? I have a T1 with 768k data and the remaining channels voice, can
the asterisk box do the Data routing + Voice processing?
Yes, zaptel will create a device node
Here's a link that will get you most of the way there:
http://www.voip-info.org/wiki/view/Asterisk+Data+Configuration
If you have any issues with setup, I recommend you contact Digium's
support to help you since I'm sure they've had the most experience
with it.
On 5/24/07, Alex Balashov [EMAIL
On 5/22/07, Asterisk [EMAIL PROTECTED] wrote:
Hello all,
One of our clients reported that they are experiencing echo in SIP calls
(even on internal ones). What do you think could be causing echo in
internal SIP calls?
We're using Polycom telephones, do you think they could be causing it?
By
On 5/22/07, Matt Scott [EMAIL PROTECTED] wrote:
Dear all.
I have what appears to be a configuration error but I cannot for the life of
me see what it is. (I am a newbie)
I have searched the wikki and google etc but still none the wiser. Any help
would be very gratefully received.
Problem:
On 5/22/07, Morgan Gilroy [EMAIL PROTECTED] wrote:
In your dial lines you have an extrac comma (,)
exten = _9xxx,1,Dial,(${OUTBOUND}/${EXTEN:1})
should be
exten = _9xxx,1,Dial(${OUTBOUND}/${EXTEN:1})
or
exten = _9xxx,1,Dial,${OUTBOUND}/${EXTEN:1}
Good catch Morgan!
On 5/16/07, Oscar Atienza [EMAIL PROTECTED] wrote:
Hi all,
I have 1 Card Digium TE412P and 2PRI E1.
I have more problems with drops lines. The asterisk log is this:
May 16 10:52:26 NOTICE[4465]: chan_zap.c:8194 pri_dchannel: PRI got event:
Alarm (4) on Primary D-channel of span 1
May 16
On 5/11/07, Juliano Fernandes Schroeder [EMAIL PROTECTED] wrote:
I'm trying to configure a TDM808P card on debian. When I modprobe wctdm24xxp
i get this error
FATAL: Module wctdm24xxp not found.
FATAL: Error running install command for wctdm24xxp
I think i have successfully compiled the zaptel
On 5/9/07, Gavin Henry [EMAIL PROTECTED] wrote:
Hi All,
What do you recommend? I was looking at:
http://www.voipon.co.uk/sangoma-a200-fxo-fxs-analogue-card-pci-express-p-393.html
But it will be 3 PCI slots.
You could do it in one slot with Digium's TDM2400P (you would actually
have to get
You may want to consider renaming daemontools as it is also the name
of a windows program that allows you to mount CD/DVD ISOs, so there
could be some confusion.
On 5/2/07, Steve Totaro [EMAIL PROTECTED] wrote:
Vicente Aguilar wrote:
Hi
I've recently released the daemontools scripts I use to
On 5/3/07, Ronaldo [EMAIL PROTECTED] wrote:
Hi all,
Is it possible to have something like this:
SoftPhone -(SIP)- Asterisk -(IAX trunk)- Asterisk -(SIP)- SoftPhone
I want a IAX trunk between two asterisks and on each tip I have SIP
clients that need to talk to each other.
Yes, Asterisk will
On 4/23/07, Patrick Fortin [EMAIL PROTECTED] wrote:
Are echo cancellation parameters useful when using the ztdummy driver and
no physical card ?
No. The echocan software and hardware only cancel hybrid echo. They
do not cancel acoustic echo that would be generated by voip phones
with bad
On 4/20/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, Apr 19, 2007 at 02:44:21PM -0700, Jay Wilton wrote:
Hello,
I'm trying to set the 3rd span of a new digium quad card as
a EM T1 for Faxes to a Hylafax server. The 1st and 2nd
spans are working as PRIs. When I start asterisk, the logs
On 4/20/07, Olivier [EMAIL PROTECTED] wrote:
2007/4/20, asterisk [EMAIL PROTECTED]:
Hi,
Does anyone know if it is possible to plug a tdm400p pci digium card
into an pci-e 16x slot ?
np
Olivier meant no here as well.
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On 4/17/07, Donovan Niesen [EMAIL PROTECTED] wrote:
I have set up a script that ensures certain services are up on my
Asterisk box (Trixbox 2.0). I would like it to trigger a wake-up call
if certain conditions aren't meant. How might I accomplish this from
the shell?
Take a look at call
Trying to find my feet here. If I wanted to connect Asterisk to a PRI and
throw in a T1 Adtran channel bank into the mix for fax machines would the
following work?
In my experience, I have never had an issue with faxes and Digium's
cards, but I'm sure many people will beg to differ.
also,
You seem to have misplaced your message/comment/question.
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On 4/9/07, Carlos Chavez [EMAIL PROTECTED] wrote:
No, that particular model is not able to use an E1. You need a TE110P
which is able to select between both. They used to have an E100P card
that was E1 only, both were replaced by the TE110P.
The TE120P is the updated version of the
On 4/9/07, Jim Freeze [EMAIL PROTECTED] wrote:
Or a new Digium TDM880B replacing the old TDM40B for only one IRQ...
Do you know if this board will fit in a 2U machine?
The TDM800P is about the same height as the TDM400P and is about an
inch longer, so you should have no problem putting it in
On 4/2/07, Klaverstyn, David C [EMAIL PROTECTED] wrote:
I have a brand new TE120P card that I have installed and asterisk is not
starting as I am getting the error: ERROR[5054] chan_zap.c: Unknown
signalling method 'pri_cpe'
Make sure you have libpri installed and that it is the right version
On 3/20/07, Mark Farver [EMAIL PROTECTED] wrote:
I suspect the issue is caused by the echo canceller, since I believe the
issue appear about the time we turned echo cancellation on (for the IAX
users). We don't need echo cancellation for PRI to PRI calls. I've
looked around, but I am finding
On 2/12/07, Paradise Dove [EMAIL PROTECTED] wrote:
my card has just fxo modules and is put in a 3.3v slot.
when running modprobe wctdm24xxp
it waits for ever and dmesg shows Resetting the modules
what could be the problem?
when i put this card in another system with 5v slot it works fine.
On 2/11/07, Joseph [EMAIL PROTECTED] wrote:
I have a few questions with regards to Digium S101I adapter.
I would like to use it as a traveling companion, plugging it into
various networks (all behind firewall I assume).
I'll be registering it into my asterisk server (behind firewall, port
4569
On 2/11/07, Paradise Dove [EMAIL PROTECTED] wrote:
does TDM2400 work on 3.3v pci slot?
Yes, all of Digium's analog cards are dual voltage and can work with
either 3.3V or 5V slots. You just need to make sure you have an extra
molex connector if you're going to be using FXS modules on the
What is the package manager used? And what is the added value compared
to the well maintained debian based asterisk ?
Hi Maxim,
AsteriskNOW is built on top of the R-Path linux distribution which
uses conary as the package manager. There is no difference between
the version of Asterisk included
On 1/10/07, Jay Moore [EMAIL PROTECTED] wrote:
Can I repeat channels like that or will it cause Asterisk to choke? If
I can't do it that way, can someone suggest a way to do it?
It will not cause Asterisk to choke, but when you assign the group the
second time, it replaces the first
If you type show application background on the *CLI, you can see all
the options listed there. The last optional argument is the context
that you want to use to look for extensions.
On 12/28/06, Time Bandit [EMAIL PROTECTED] wrote:
I am using the Background() function to ask for the
Giorgio,
I believe the syntax for mISDN is mISDN/port:channel/number. In other
words, replace your - with a :.
On 8/25/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Hi,
I have a quadBRI beronet ISDN card. Is there anybody who knows how to
choose the channel to make calls? I tried with
However, reinstalled the box from ground up and installed 1.2.10 and now
CLID isn't working at all. The PSTN line is still transmitting it, as
I've plugged in my Uniden cordless with CLID and it shows up fine on
there, but getting absolutely nothing inside the ${CALLERIDNUM} and
${CALLERIDNAME}
On 8/18/06, David Cook [EMAIL PROTECTED] wrote:
Maybe I'm daft, but can asterisk to 'or' logic in dialplan matches sort
of like the SPA's can?
Tollfree numbers for example. I can have a line for each combination:
exten = _1800NXX, Dial,
exten = _1866NXX, Dial,
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