hmmm.. no ideas?! :-|
tom
Thomas Artner wrote:
Hi!
At the moment i am using a digium tdm400 card for my analog phone lines.
The zaptel driver supports fax detection, so incoming faxes are
redirected to the fax extension automatically.
This works without problems with asterisk 1.2
Hi!
At the moment i am using a digium tdm400 card for my analog phone lines.
The zaptel driver supports fax detection, so incoming faxes are
redirected to the fax extension automatically.
This works without problems with asterisk 1.2.
But now I would like to switch to ISDN (mISDN) and asterisk
Hi!
Either the fax machine or the asterisk box has to pick up the call to
know whether it is a fax or not.
My solution is that I let asterisk pick up every call, and if it is a
fax, then the call is forwarded to a fax-machine.
If its a voice call, the call is forwarded to the phones.
Hi!
I would like to connect a door phone to my asterisk server.
I decided to do that with an analog (a/b) doorphone and a sipura box.
Can anyone give me a recommendation for a door phone with good voice
quality?
I am from Austria/Europe and I looked at products from Rocom and from
Auerswald.
It depends on the actual given environment, but you could also think
about using Linksys' PAP2 adapter!
mike wrote:
Dear list
which hardware solution would you suggest for connecting 60 analog
phones to asterisk ?
thank you very much
.mike
Hi!
I have an analog fax machine connected to a sipura ATA which is connected to
my asterisk box.
In my asterisk box I have a digium card for a connection to the public
telephone network (analog).
On this digium card is also a us robotics sportster modem (analog) connected.
My problem:
I can
Hi!
I can't find the link anymore where it was a statement from the nokia
support that they are working on a STUN implementation.
A firmwareupdate (with STUN support) will be available in fall 2006.
tom
Andreas Sikkema wrote:
Anyone here use the Nokia E61 ? I am looking to invest in a
wifi
here is the link (german):
http://www.my-s60.com/de/news/news/newswann_kann_die_e_serie_nat_traversal/back/105/cHash/bac3645f5e/index.html
STUN will come in fall 2006
TURN and ICE in 2007
Thomas Artner wrote:
Hi!
I can't find the link anymore where it was a statement from the nokia
joea, j4computers wrote:
As a complete newcomer to Asterisk, Digium and PBX, I have several questions.
But I'll start with this.
To setup a simple system with only a couple of POTS lines, I gather I will
need a TDM400 board with FXO and/or FXS modules.
So, a TDM400 card will support up
Hi!
I am working hard on getting a useful attented transfer. (The built-in
atxfer feature isnt useful - because of calls getting lost - has been
discussed a few months ago)
I have all my analog phones on sipura boxes. With the flash hook i can
do such attended transfers without problems now.
Thomas Artner wrote:
Hi!
I am working hard on getting a useful attented transfer. (The built-in
atxfer feature isnt useful - because of calls getting lost - has been
discussed a few months ago)
I have all my analog phones on sipura boxes. With the flash hook i can
do such attended
Damien Gabrielson wrote:
Hi,
I'm looking for a simple way to send email from a dial plan. I have
searched around quite a bit looking for a solution for this and I'm
surprised that I haven't found anything useful yet other than using the
System() application. I would like to be able to
Hi!
I have connected my analog phones to an asterisk box with sipura spa2002
devices.
I can do an attended transfer by taking the call which should be
transferred, pressing the flash button, dialing the number to which the
call should be transferred and now i can hang up or talk to the person
who
Alexander Lopez wrote:
This might be what you're seeking;
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ChanIsAvail
If the phone rings, then the channel IS available. The solution is to
disable call waiting on the SIP device.
The s option needs to be used:
s - Consider the
Hi!
I have two extensions (25 and 26, and so two phones) for one person in
an office.
I can dial 25 or 26 and always both extensions are ringing. Thats okay!
exten = 25,1,Dial(Sip/25Sip/26)
exten = 26,1,Dial(Sip/25Sip/26)
The problem with this solution is, if the person is talking on one phone
Hi!
I've one card over from my last asterisk project.
The card is about 3 months old, a copy from the invoice for warranty is
available.
Location: Vienna, Austria.
If anyone is interested - send me a private mail.
cheers,
Tom
(i hope this mail is okay for this list)
Am Wednesday 26 April 2006 20:43 schrieb Wai Wu:
If I download zaptel-1.2.5, do I still have to apply the
zaptel-1.2.5-patch?
no.
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hmm.. does really nobody had such an issue before?
Thomas Artner wrote:
Hi!
I am using asterisk with two tdm400p cards.
Sometimes (one call out of ten), when a call comes in and is taken,
there is some terrible noise for a short time in the line (for about a
second).
Both partys can
Hi!
I am using asterisk with two tdm400p cards.
Sometimes (one call out of ten), when a call comes in and is taken,
there is some terrible noise for a short time in the line (for about a
second).
Both partys can hear the noise. And sometimes the call has to be hung
up, because the noise doesn't
Hi!
I decided to open an issue about this case in the mantis database!
I am not very familiar with the bug/issue tracking procedure at the
asterisk project, but I think i can make it.
Is there something that would speak against it?
cheers,
tom
Thomas Artner wrote:
Hi!
A few months ago
here's the reported issue: http://bugs.digium.com/view.php?id=6973
cheers,
tom
Thomas Artner wrote:
Hi!
I decided to open an issue about this case in the mantis database!
I am not very familiar with the bug/issue tracking procedure at the
asterisk project, but I think i can make
Hi!
After upgrading to asterisk 1.2.7.1 (from 1.2.4) it seems that app_rxfax
doesnt work any more.
I've installed spandsp-0.0.2pre25 (the same problem with
spandsp-0.0.3pre6.tgz ) app_rxfax.c, app_txfax.c and made the Makefile
patch from the same directory.
When starting asterisk I always get
Rob Terhaar wrote:
did you try to recompile the plugin?
yes, of course...
On 4/14/06, Thomas Artner [EMAIL PROTECTED] wrote:
Hi!
After upgrading to asterisk 1.2.7.1 (from 1.2.4) it seems that app_rxfax
doesnt work any more.
I've installed spandsp-0.0.2pre25 (the same problem
after a few hours of debugging it works now...
I got some version mixes of spandsp on my system...
sorry for the spam
tom
Thomas Artner wrote:
Hi!
After upgrading to asterisk 1.2.7.1 (from 1.2.4) it seems that app_rxfax
doesnt work any more.
I've installed spandsp-0.0.2pre25 (the same
Hi!
A few months ago I needed some help for the following issue:
.) a call comes in
.) Person A takes the call and does an attended transfer to Person B
.) Person A hangs up the phone without waiting for Person B taking the call
.) the caller get lost at this point !!
At this point the
Michael Collins wrote:
A few months ago I needed some help for the following issue:
.) a call comes in
.) Person A takes the call and does an attended transfer to Person B
.) Person A hangs up the phone without waiting for Person B taking the
call
.) the caller get lost at this point !!
Melcon Moraes wrote:
So, what version of spandsp are using afterall?
i am using spandsp-0.0.2pre25 now.
In the 0.0.3 package, there is no app_rxfax.c and no app_txfax.c. No
idea why thats missing there.
tom
[]'s
MM
-Original Message-
From: Thomas Artner [EMAIL PROTECTED
Tomislav Parčina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
you are using the attended transfer feature..
ist it already possible to hang up before the other person lifts the handset
without loosing the caller when you are doing an attendet transfer?
(person A takes an
Am Tuesday 14 March 2006 18:38 schrieb Barry Flanagan:
Hi,
We are trying to use attended transfer with Asterisk 1.2.5, but when we
do the transfer and dial the new number, it times out after 3 rings and
then the callee is put back to the original agent.
Where can I adjust the timeout which
Am Monday 27 February 2006 23:15 schrieb Anton Krall:
Guys.. I just thought of something.. Anybody who is sucessfuly receviing
faxes using spandsp and running Fedora Core 3?
What are you running?
Debian stable - and it works perfectly.
|-Original Message-
|From: [EMAIL PROTECTED]
Anton Krall wrote:
Ok 1 for Debian, any Fedoras Core 3 out there?
I think it doesn't depend on the linux distribution whether it works or not.
It's rather an hardware issue.
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Thomas Artner
Am Saturday 25 February 2006 23:49 schrieb Anton Krall:
Whats mpack tom?
a command line tool for easily sending emails with attachments.
I use sendEmail..
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Thomas Artner
|Sent: Saturday
Am Saturday 25 February 2006 19:38 schrieb Steve Underwood:
Cosmin Prund wrote:
I've noticed some other odd thing with rxfax. In my case I can receive
faxes (using TDM400P) just fine. I can only see those faxes using Windows
XP's Fax and Picture thingy, other applications are having trouble.
: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Thomas Artner
|Sent: Saturday, February 25, 2006 12:47 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] fax receive using TDM400P
|
|Am Saturday 25 February 2006 19:38 schrieb Steve
:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Thomas Artner
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: Wildcard TDM400P REV I (4 modules)
...
Zaptel Version: 1.2.4 Echo Canceller: KB1
maybe it depends on different hardware revisions?
i don't know...
tom
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Thomas Artner
|Sent: Friday, February 24, 2006 8
Am Friday 24 February 2006 21:17 schrieb yrving rivas:
Thomas: does it work in your case?
Do anybody have the fax working w/tdm?
Yes, as I wrote before, receiving faxes with a tdm400p card works perfectly!
Thomas Artner [EMAIL PROTECTED] escribió:
Am Friday 24 February 2006 16:48
Am Tuesday 21 February 2006 00:24 schrieb Marc Archer:
Hi All,
Can someone give me a definite answer as to wether or not you can
reliably run multiple TDM400P's in the same machine?
I need 4 x FXO and 4 x FXS to connect to both the PSTN and existing key
system, but I have seen several
PROTECTED] On Behalf Of Thomas
Artner
Sent: Sunday, February 12, 2006 9:39 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] asterisk + door opener
Hi!
I am new to asterisk and I'd like to know wheter the following scenario
is possible:
Someone press the Button
Hi!
I am new to asterisk and I'd like to know wheter the following scenario
is possible:
Someone press the Button on the door station.
The door station dials lets say the extension 333.
I take the call on 333 and talk with the person on the door.
Now I'd like to activate the door opener by
Questions for the community: is an integrated transfer feature
valuable to you?
Yes, merging blind and attended transfer would be valuable for me!
If so, would you be willing to put out a bounty?
Maybe. Depends on how much it would be.
Tom
Hi!
I got this answer from the digium support:
You may wish to use the attended transfer by either using hold or
flashhook instead of the # features.conf attended transfer option. From
a phone connected via a Zap channel, you would need to hit flash. Then
enter the extension which you wish to
Hi!
I am new with asterisk and I have my first problem with the attended
call transfer feature.
When a call comes in, i take the call and i would like to transfer it.
So I press the * button (mapped for the attended transfer in
features.conf) and the number for the receiving extension.
The
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