?
Thanks,
Cosmin Prund
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anyway to display
the actual talk time somewhere. It might also be that I'm an programmer and I
attempt to solve all problems writing programs, so maybe someone else has a
better idea!
--
Cosmin Prund
De la: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] În numele Douglas Garstang
Trimis
now got 99,95 average timing from
zttest with the worst being over 99,00. You might want to try this yourself
since you also seem to have the X (pci express) version of the card.
--
Cosmin Prund
De la: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] În numele Jeremy Mann
Trimis: Wednesday
to give them a
lot of information about the calling clients before they actually take the
call!
P.S: I'm not an Asterisk guru. I'm not in the business of selling Asterisk. My
experience comes from implementing Asterisk for my own organization - so it's
limited experience.
--
Cosmin Prund
and music has pauses. One can imediatelly tell the
ringing is done but they might need a few extra seconds to realise the music
has stoped. On the other hand the client has no such problem since he/she hears
ringing just before they get bridged to the MeetMe room.
Any ideas? Thanks!
--
Cosmin
in writing an softphone that emulates the
Skype API?
Should I just give up and buy a Siemens DECT phone that supports a
bluetooth headset?
--
Thanks,
Cosmin Prund
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asterisk
.
Out of the three, the first option is probably best (I'm obviously no
Asterisk-guru). (b) might work well as I might be able to actually loop till I
send the fax, not till I get an answer; (c) might also work, but I sure hope I
don't need to go there...
--
Cosmin Prund
use of a bluetooth headset,
free or comercial?
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Thanks,
Cosmin Prund
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the
local channel. Besides, looping in the dialplan would probably
interfere with the Wait option in the .call file so that's a really
bad solution.
--
Thanks for any suggestion,
Cosmin Prund
___
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Behalf Of Anselm Martin Hoffmeister wrote:
Subject: Re: [asterisk-users] About .call files when the congestion is
on myside
Am Montag, den 15.10.2007, 11:26 +0300 schrieb Cosmin Prund:
Hello everyone.
I’m working on an application that needs to automatically send faxes.
To send
NAT is not that big of a problem, not anymore.
Do a NAT search on http://www.voip-info.org - it'll get you started (got me
started at least)
--
Cosmin Prund
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Henry Cobb
Sent: Tuesday
on two dual core amd machines and
i'm not getting timing issues on three dual-core intel machines.
--
Cosmin Prund
-Original Message-
From: Adam Robins [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: 29.05.07 18:09
There was a discusion on the subject a few days ago, search the archives . The
quick answer is you don't, but don't take my word for it, I know nothing
about xen and very little about asterisk!
--
Cosmin Prund
-Original Message-
From: Roberto Pereyra [EMAIL PROTECTED]
To: Asterisk
It just works. Simply transfer your call to the desired extension and
let Asterisk take care of the details.
--
Cosmin Prund
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of DiegoF
Sent: Friday, May 25, 2007 12:04 AM
To: Asterisk Users Mailing List - Non-Commercial
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Remco Post
Sent: Tuesday, May 22, 2007 11:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Working softphone for poket PC
Cosmin Prund wrote:
Googling arround I found a number
cool.
Guess I agree with you, this is another men's toy, since it really has
no practical use to it.
--
Cosmin Prund
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anselm
Martin Hoffmeister
Sent: Tuesday, May 22, 2007 11:08 PM
To: Asterisk Users
for sure they work really well).
Specifically: Does the Siemens support rejecting an VoIP call? Does it
start ringing showing Caller info immediately (as opposed to the
ATA+DECT combination, as that requires 1 ring to get caller info)
Thanks,
Cosmin Prund
-Original Message-
From: [EMAIL
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Remco Post
Sent: Wednesday, May 23, 2007 10:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] WiFi SIP phones
Tony Plack wrote:
Are the
Googling arround I found a number of pocket pc softphones. Of those I was only
able to install SJ-something (removed it).
Is there one (pocket pc softphone) that works?
Thanks,
Cosmin Prund___
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I'm no Asterisk expert (nor Linux expert) but I am using my * box for
multiple things (transparent firewall, NAT box, samba server, poptop server)
and for a considerable time I've been running a VmWare server with a Windows XP
virtual machine up-and-running at all times! The Windows XP VM was
through the code possible.
--
Cosmin Prund
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nick
Seraphin
Sent: Tuesday, May 15, 2007 9:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] [*Win32 0.60] Sending call
I've managed to configure a SPA3012 to do that a few days ago. I
remamber using something like S0:[EMAIL PROTECTED] for the
#1 dial plan. Unfortunately I no longer have access to the SPA because I
shiped it to an co-worker and this co-worker didn't manage to install
it yet.
I also remamber an odd
Are you using automatic rules for sorting email?
My Outlook 2007 miss-sorted all of those announcements (for Asterisk 1.4.3 and
a few others - that's the only one I cared about) and I missed them. Maybe your
mail roules are just as wrong.
--
Cosmin Prund
-Original Message-
From
in the
Dialplan, or you may use an AGI to consult a CallerId database (plain
text file).
On the other hand, if you've got your AGI going, stick to that!
--
Cosmin Prund
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Troy Ayers
Sent: Thursday, April 26, 2007
How about connecting the analog phone in the elevator to an ATA gateway
that provides PSTN FallThrow? Sipura 3000 or Lynksis SPA3102 can do
the trick.
--
Cosmin Prund
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erik
Wartusch
Sent: Thursday, April 26
you DON'T have softdmtf=yes or relaxdtmf=yes in
your
capi.conf.
--
Thanks,
Cosmin Prund
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#0x4bbf LEN=0012
Controller/PLCI/NCCI= 0x301
-- ISDN1#02: info element DISCONNECT
-- ISDN1#02: Disconnect case 3
-- CAPI queue frame: TYPE: Control (4) SUBCLASS: Hangup (1) ]
[ISDN1#02]
/CLI Output
--
Thanks,
Cosmin Prund
-Original Message-
From: [EMAIL
IVR! Is there
any way to enable/disable DTMF clamping on a per-call basis? Or better,
disable DTMF only when the call makes it to an operator?
--
Thanks,
Cosmin Prund
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Armin Schindler
Sent
the softftfm setting in capi.conf, no luck
ether way.
Is there anything else I can try? Did I miss the obvious (it would not
be my first)
--
Thanks,
Cosmin Prund
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Armin Schindler
Sent: 20
or the other
party accidently touches the dial pad on the phone).
Does one of Asterisk, chan_capi or the Diva driver have support for such
functionality?
--
Thanks,
Cosmin Prund
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I've had similar behavior on my own IVR. I moved my sound files to a ram
disk and all pops and ticks stopped!
David Brazier wrote:
Hi All
I have a problem with IVR scripts which consist mainly of Playback of
audio files, driven from an AGI application. There are clicks every few
seconds or
I don't think it works. I tried calling my own yahoo messenger ID with
no success: it rings a number of times and then it goes to some sort of
voice mail.
And I did invite the user they specified to my yahoo list, I also
entered my yahoo id into the registration form on the site.
I used a
The network terminator installed by the Telco in Romania works the same
way: it has two analog outputs and two digital (S0) outputs. I've also
got a full TDM400 card with 3 FXS and one FXO, but I gave them up gladly
for a proper ISDN card (I'm using a Diva Eicon Server) - and I don't do
: Digium support responded very quickly.
Cosmin Prund wrote:
Hello: I got into a trap. As far as I know I do not need to pay any
royalties to use G.729b in Romania, so I should have used other
drivers. The installation procedure looked difficult so I decided to
get one from Digium - it's
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Cosmin Prund
Sent: Tuesday, February 06, 2007 6:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] New Issue
Try it like this:
cd /usr/src/asterisk-1.4.0
make clean
./configure
Hello: I got into a trap. As far as I know I do not need to pay any
royalties to use G.729b in Romania, so I should have used other drivers.
The installation procedure looked difficult so I decided to get one from
Digium - it's not that expensive, my time is much more expensive.
Made the
So simple... I'm doing that right now, I've sent them an email.
I didn't find that email address on Digium's support page...
Thanks.
Bruce Ferrell wrote:
Cosmin Prund wrote:
Hello: I got into a trap. As far as I know I do not need to pay any
royalties to use G.729b in Romania, so I should
Let's see if I remember this, it gave me a bit of trouble as well.
*after* you made sure you've got the zaptel driver in order, go to the
src folder for asterysk and issue make menuconfig.
Go to 3 and see if you have the chan_zap listed there and with [*]
prefix. If it's not listed it's because
I'v got Asterisk 1.4.0 and it understands make menuconfig. Is your
version older or newer? If it's older, maybe you can try the newer one.
If it's newer - I'm out of ideas.
David Ruggles wrote:
Thanks for the reply, but when I go to the asterisk source directory and
issue make menuconfig I
Try it like this:
cd /usr/src/asterisk-1.4.0
make clean
./configure --with-zaptel=/usr/src/zaptel-1.4
make menuconfig
make all
make install
David Ruggles wrote:
Sorry about that I must have been in the wrong directory. I also have 1.4.0
and I tried it again and it worked. Chan_zap is not
Thanks, it really was easy.
Unfortunately it only works for MSN's, not for the base number. Oh
well, I'll just stop using the base number, I've got enough MSN's anyway.
Thanks again.
Armin Schindler wrote:
On Thu, 1 Feb 2007, Cosmin Prund wrote:
Any ideas? It should be simple
Armin Schindler wrote:
What type of line is that? The 'base number' is also a MSN on lines I know.
Or is it PtP with DID?
Armin
On Sat, 3 Feb 2007, Cosmin Prund wrote:
The base number works like any other MSN most of the times, but the
busy application doesn't work on it. If I dial
Tzafrir Cohen wrote:
Do you use Busy to send a bus signal to the other party?
I use Busy. I have no idea how it works. When I call from my mobile
phone to my PBX I get a busy signal and it seems I'm not being charged
for the call (so it's not like * opened up the line and played the
Any ideas? It should be simple...
Cosmin Prund wrote:
Hello everyone:
using chan_capi 0.7 and asterisk 1.4
Quick question:
How can I detect the number of voice channels (B channels) in use at
a given time. I'd like to call Busy if two B channels are used on my
BRI to give the calling
) In a distant future I
might use the capi channel's ability to transfer the call to a different
POTS line since this doesn't use the B channel.
Thanks,
Cosmin Prund
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switch doesn't do standard VLan, it only does
VLan if linked to an other Trendnet switch - not useful at all!
Thanks,
Cosmin Prund
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is flowed and needs
to be replaced.
Thanks,
Cosmin Prund
Patrick Cervicek wrote:
Cosmin Prund schrieb:
P.S: For those that don't understand WHY I can't trust marketing
material, let me tell you something about the Trendnet switch that's
fast becoming garbidge. I wanted an managed switch so I boght
/src/asterisk-1.4.0/include
MODULES_DIR=/usr/lib/asterisk/modules
CONFIG_DIR=/etc/asterisk
//--
If anyone has any idea what I'm doing wrong, please help me,
Thanks,
Cosmin Prund
This gets fixed using the chan_capi.HEAD, not chan_capi.0.7.1; chan_capi
compiled right out of the tar.gz, no changes required (the defaults in
the Makefile are ok)
Cosmin Prund wrote:
I've got a brand new Eicon Diva Server BRI card and I want to
configure it with Asterisk. I managed to get
less then $1000 / 750 euro
(b) Has one or (preferably) two ISDN S0 interfaces.
(c) Easy to set up.
(d) Drivers offer proper echo-canceling OR has an hardware echo
canceler. I might increase the $1000 a bit if I can get good hardware
echo canceler...
Thanks,
Cosmin Prund
I do not care about fax. I just want a good VOICE card.
Can someone please give a price quote for this card, give or take 10%? I
just spent 5 minutes filling in a really long form on a shopping web
site to get a price quote, only to find my account needs to be manually
activated before I can
How about the Digium Wildcard B410P card? It seems to be Digium, it
has hardware echo cancel and I can buy this in Romania. Is this card any
good?
Cosmin Prund wrote:
Hello everyone.
I need a BRI ISDN card that works in Romania. I already have one of
the Cologne HFC-S PCI cards
I finally found a price tag for the darn thing, at around 500 euros I
can handle it.
Qustion: Do they behave properly if I've got an other Digium TDM400 card
in the system? How about installing two cards in the same server?
At the moment I've only got 1 ISDN line plus a few analog lines going
Following a discussion on this list about a week ago I downloaded and
installed Debian Linux. Now I want to install asterisk-bristuff.
How do I do that?
Better yet, what do I put in /etc/apt/sources.list so I can do
apt-get install asterisk-bristuff
--
Thanks for your help,
Cosmin Prund
Linux. Now I want to install asterisk-bristuff.
How do I do that?
Better yet, what do I put in /etc/apt/sources.list so I can do
apt-get install asterisk-bristuff
--
Thanks for your help,
Cosmin Prund
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version of bristuff). I'm hoping for a quick
answer like: Install LinuxVariant 10.20, install all updates using
LinuxVariantUpdateProgram, download bristuff version X.Y.Z, call
install.sh and be done with it.
Thanks,
Cosmin Prund
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I know, but I'm willing to use anything as long as it gets
the work done.
Thanks again,
Cosmin Prund
Filip Drągowski wrote:
Bristuff 0.2.0-RC8 if for Asterisk 1.0.10
Bristuff 0.3.0-PRE-1r if for Asterisk 1.2.9.1, libpri 1.2.3 and zaptel
1.2.6
download proper versions
(for Asterisk 1.2.9.1
I don't use Debian but I'm going to give Debian a try. I'm downloading
the Debian distribution right now. Unfortunatelly it will be about a
week till I'll download the whole 8Gb (2xDVD iso).
Tzafrir Cohen wrote:
On Wed, Jul 19, 2006 at 03:17:15PM +0200, Filip Drągowski wrote:
I'm using a HFC-S ISDN card and a TDM400P zaptel card with 3xFXO and
1xFXS modules.
How did you finally manage to compile bristuff on Centos 4? I'm
downloading Debian right now but the 2 DVD images will take about a week
to download so I'm willing to try anything else in the time. I've got
Tzafrir Cohen wrote:
On Wed, Jul 19, 2006 at 09:30:30PM +0300, Cosmin Prund wrote:
I don't use Debian but I'm going to give Debian a try. I'm downloading
the Debian distribution right now. Unfortunatelly it will be about a
week till I'll download the whole 8Gb (2xDVD iso
Your best bet is to write it by hand. If you don't want that you may use
an AGI script to interrogate the operating system, selecting the one IP
address you like the most. Please take into account system's with more
then one net card. Please also note you may have more then one IP
address per
wrote:
On Tuesday 23 May 2006 14:54, Cosmin Prund wrote:
Is there a known hack or patch to blow Asterisk's echo canceler up to
128ms? Or at least 64 ms?
It's real easy... at least for us code tinkerers.
It's even easier now, as Digium has just committed these changes to svn
Andrew Kohlsmith wrote:
Regarding echo tail: Asterisk's default software echo cancellation is only
32ms at maximum. You can blow this number up to 128ms (where the hardware
echo cancellers sit at), but it costs more in CPU time and memory, and I
think that Digium left it at 32ms as a
Are you doing something funny with the CID on it's way to the phone?
I've got a somewhat similar problem with an Aastra IP phone (yes, I did
say IP): it would NOT ring if the caller id started with an #. Maybe
your Aastra PSTN phone got some of the same (buggy?) handling of CID's?
Dan Elder
Take a look on the Digium site for the requirements of the card. Most
likely the old computer can't handle the card.
M.Hockings wrote:
I just bought an TDM400P card with one FXS port and a X101P FXO card
to try and put together the beginnings of a PBX here. The computer
they are going in
I wanted to see where those periodical spikes are coming from so I
started shutting things down. The first thing to go was Asterisk. Next
went many different things like mysql and ntpd. Finally I killed zaptel
(/etc/init.d/zaptel stop) - and the spiking stoped!
Next I rebooted and I've done
I'll give this one a try, but don't trust me, test it yourself :-)
Of course Asterisk can do it! All you need to do is set up a rule for
matching ALL extensions in the PBX in it's own separate context and
include that context into your normal context. In the following
example, asterisk will
I'm fairly new to Asterisk myself and I also started with AAH.
Unfortunately I had to remove all configuration files generated by
FreePBX (the GUI of AAH) and started over using http://voip-info.org as
my guide. Configuration files generated with FreePBX make use of
advanced functionality
Ralf Schlatterbeck wrote:
On Fri, May 12, 2006 at 02:25:44PM +0300, Cosmin Prund wrote:
Unfortunately the latest misdn-mqueue does not compile on my system, it
issues all sorts of blah-blah that I'm interpreting in only one way:
there's a problem with the parameters the Makefile passes
Hello everyone.
I've got a HFC ISDN card that I'm using with chan_misdn and it basically
behaves like crap. Echo is waaay worst then echo I get TDM400 card,
sound is choppy (there other side is allays complaining about sound
interruptions) and to top it all it detects fake DTMF's all the
Woodoo People .pGa! wrote:
I've got a HFC ISDN card that I'm using with chan_misdn and it basically
behaves like crap. Echo is waaay worst then echo I get TDM400 card,
sound is choppy (there other side is allays complaining about sound
interruptions) and to top it all it detects fake DTMF's
Chris Bagnall wrote:
I've got a HFC ISDN card that I'm using with chan_misdn and
it basically behaves like crap. Echo is waaay worst then echo
I get TDM400 card, sound is choppy (there other side is
allays complaining about sound
interruptions) and to top it all it detects fake DTMF's all the
Hello everyone. I've got this really annoying HFC Cologne card (or
however I should call it - a single channel ISDN card based on the HFC
chipset).
It wrongfully detects lots and lots and lots of incoming DTMFs, to the
point the card is not usable.
Here's a sample out of CLI:
P[ 1] I IND
You can't possibly read conf files and present them to the user in a
web-based form because of the complexity of the data in those file. Take
the extensions configuration file for an example (as this is probably
the most complex one). Settings in this file might specify a simple
mapping
I'm using ChanSpy. Set up an extension with ChanSpy, dial the given
extension and don't hang up (put it on speakerphone). When there's no
one on the given zap channel you'll here silence. As soon as someone's
on the channel you'll be listening to them.
If you don't like ChanSpy there's
Hello everyone.
What's better for Asterisk: have 2 distinct 100Mb network cards in the
system, one on the internet and one on the local net OR have one
1000Mb network card with 2 separate VLAN's set up? It's a difficult
decision because 2 cards are using 2 IRQ's etc but a single 1000Mb card
!
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Cosmin Prund
Sent: Tuesday, April 25, 2006 10:41 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Help on chan_misdn and MSN's
Quick question:
Is there a way
Quick question:
Is there a way to distinguish between calling MSN's when using chan_misdn?
More info:
I've got my ISDN2 (EuroISDN) up and running here in Romania with 1 base
number plus 5 MSN's. Now I want to my * to do different things when
receiving a call on from different MSN's (like
around and ended up into an arbitrary context.
Thanks for any help,
Cosmin Prund,
Romania
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From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of C F
Sent: Monday, April 03, 2006 3:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] call transfer to external phone number
Yes, as long as the context that the
Hello everyone.
Is it possible to do some very basic voice recognition from within
Asterisk's dialplan? What I'm aiming at is the ability to speak the digits I
want to dial from my mobile phone. Dialing digits on my mobile phone while
driving is not all that safe...
Thanks for any input,
Cosmin
-Users] Coice recognition IVR?
Hi there
Sphinx does speech recognition:
http://www.voip-info.org/wiki-Sphinx
HTH,
Cristi
On 4/3/06, Cosmin Prund [EMAIL PROTECTED] wrote:
Hello everyone.
Is it possible to do some very basic voice recognition from within
Asterisk's dialplan? What I'm
recognition IVR?
Cosmin Prund wrote:
Hello everyone.
Is it possible to do some very basic voice recognition from within
Asterisk's dialplan? What I'm aiming at is the ability to speak the
digits I
want to dial from my mobile phone. Dialing digits on my mobile phone
while
driving
Also be aware Asterisk is probably runing
in its own, non-root account. It needs execute access to the program,
and you need to specify full path. At least thats what worked for me J - dialing 500 on my box
does System(/sbin/reboot) !
From:
[EMAIL PROTECTED]
[mailto:[EMAIL
Hello everyone.
My land-line provider (Romtelecom) has a very nice offer for ISDN. All in
all they offer me a digital land-line with 1 base number + 2 MSN's and that
would make a grate addition to my full-time home office.
Romtelecom say they're providing EURO-ISDN and the line is compatible
The line is supposed to be standard EUROISDN; I found no mention of DSS1 on
the technical specs page, I only found EUROISDN.
The TELCO is going to provide me with a NT equipment that has two analog
ports and two S0 ports; Of the two S0 ports one is supposed to be used to
connect the PBX to the
Thanks for the tip!
I shoud have found this on my own...
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of C F
Sent: Friday, March 03, 2006 5:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
, 2006 7:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Changing caller id on transfer
On 03/03/06 04:17 Cosmin Prund said the following:
How can I change the caller id on a transferred call so the called party
knows the call has been
AFAIK there are problems with repeatedly connecting and disconnecting the
manager interface. Also you're probably using a proxy (all manager
interfaces I've seen are using proxies), it might not be a good idea to pool
something out of the manager that often.
Did you consider running a cron job on
As usual, this is most likely a easy question, but here it goes any way:
How can I change the caller id on a transferred call so the called party
knows the call has been transferred from a colleague and it's not coming
directly from our outside lines?
The story goes like this:
1) Client calls.
I'm no Asterisk genie but if you're running on one server you're probably
not dealing with lots of users (for what I'm trying to say 100 users are not
a lot of users). Factoring in the VERY simple format of both sip.conf and
extensions.conf, isn't it possible to create an php page that would
in and gets bridged, because the code is
exactly the same except for setting callerid.
Hope that helps more than it confuses.
Joseph Tanner
On 3/2/06, Cosmin Prund [EMAIL PROTECTED] wrote:
As usual, this is most likely a easy question, but here it goes any way:
How can I change
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Matt Riddell [NZ]
Sent: Thursday, March 02, 2006 10:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Changing caller id on transfer
A thread on running 5000 simultaneous cllas ran on this list recently and it
did generate a lot of heat. You might want to look it up the archives - but
make sure you read as many posts on it as possible because lots of different
opinions formulated over time.
-Original Message-
From:
How about this:
--- Results after 33 passes ---
Best: 100.00 -- Worst: 99.987793 -- Average: 99.988163
Faxing is working just fine. Mabe it's mother board related?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent:
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. Also
printing those faxes is a bit odd: the preview is just fine but I always
need to
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.
Also printing those faxes is a bit
Windoes'es DEFAULT image viewer also has problems showing rxfax-generated
TIFF's. They do show up properly on screen but when printed the orientation
needs to be changed. I don't know if MS's viewer is somehow broken OR the
tiff is somehow broken (also I don't care) but viewing rxfax-ed tiffs is
Hellow everyone, here's an other newby question.
I've got a * configured with the card in the subject line. At times Asterisk
fails to notice a disconet from the incoming line going into one of the FXO
ports. Consequently it just keeps the line off-hook for ever and that causes
my provider to
I'm a newbye myself so beware!
(1) http://www.voip-info.org is your friend. I've got most of my info off
that site and it's a good place to start.
(2) Download a Softphone like XLite (you'll also find info on softphones on
voip-info) and start experimenting on site. When you'll be able to
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