Mark,
In the time it took to write all that you could probably have read up
enough about T.38 to realise you were talking complete rubbish :-)
Regards,
Steve
Mark Eissler wrote:
On Feb 25, 2005, at 10:20 AM, Lee Howard wrote:
In a traditional analog fax you have modulated audio data, that is,
wants to suggest corrections or additions, just blurt them out.
Regards,
Steve
Rich Adamson wrote:
Steve Underwood,
Would you mind summarizing where/how T.38 functions, and maybe how it
compares to the analog fax environment for the asterisk-users arhives?
Seems to be some misunderstanding, and a lot
Robert Terzi wrote:
I found an old Dialogic card in an abandoned PC, that I think is a
Dialogic D/41EPCI based on some googling.. The lspci output says:
00:09.0 Bridge: PLX Technology, Inc. PCI - IOBus Bridge (rev 01)
Subsystem: Dialogic Corp: Unknown device 0529
I'm just getting started
Lee Howard wrote:
On 2005.02.27 11:28 Jon Gabrielson wrote:
You wouldn't happen to know how to do this would you?
I currently have a box with both hylafax and asterisk installed.
asterisk handles the dedicated voice lines over a t100p and
hylafax handles the dedicated fax lines over a 4port serial
Lee Howard wrote:
On 2005.02.27 09:30 Martijn van Oosterhout wrote:
On Sun, Feb 27, 2005 at 09:10:48AM -0800, Lee Howard wrote:
Fax cannot handle a one-second delay. As Steve mentions in the
article, per-spec fax has some timings (particularly silence in
direction switching) set at 75 ms +/-
Peter Svensson wrote:
On Mon, 28 Feb 2005, Edwin Groothuis wrote:
For the project I've used the Eicon DIVA card. It has 8 BRI ports,
and for about 25% of the time there are 7 or 8 in use. So we want
to replace it with an E1 card. Only issue is, replace it with what?
The idea we have been
Olle E. Johansson wrote:
Steve Kann has developed a new jitterbuffer for IAX2, that hopefully
will be integrated into Asterisk v1.1 soon, to be part of the 1.2
stable relase.
Zoa and his bulgarian team is porting this buffer to SIP/RTP, but
needs support in the form of funding in order to take
Daryl G. Jurbala wrote:
I'm looking for an application that can monitor a channel for voice
input and then proceed on. The closest thing I've found is
BackgroundDetect, which expects DTMF.
Here's what I'm doing:
-Call file generated which calls someone and connects them to an
extension.
CClarke wrote:
Hello All ~
Having problems sending and receiving faxes with SpanDSP. I am testing on a
simple 2 analog POTS to 2x X100p set up, connecting one line to a Konica 720
fax machine to test, or with other remote fax machines. Voice calls are
working pretty well now. Platform is
Dennis Webb wrote:
This seems to be how AGGRESSIVE_SUPPRESSOR works. To make sure you
don't get echo, it does what a speakerphone does, mute the other party
if it hears audio from your end. There is a setting in mec2_const.h
for AGGRESSIVE_HCNTR=160 that says in the comments 20ms, I'm
Steve Kann wrote:
What he describes is echo suppression. Because an echo canceller can,
generally, only remove some part of an echo, not the entire echo,
systems are generally designed to suppress the residual echo in some
circumstances. Old speakerphones had poor on no echo cancellation, so
Hi John,
You didn't say what kind of cellular system. If its an AMPS system (I
don't think any other analogue cellular stiff exists) DTMF is quite
troublesome. If it is a digital network the DTMF actually comes from the
basestation, rather than the phone. Its is normally very high quality.
Leo Ann Boon wrote:
Has anyone on this list gotten hold of these cards? It's been 2 months
since their official ship date.
Even the website www.ipvolution.com is in wee-wee land.
It has been down for several weeks. The cards are still shown on
www.atacomm.com. I don't know whether that is a
Hi,
My guess would be the lengths in the header are not set right. If a wave
file (or a file with a similar structure, like TIFF) works with some
things and not with others, the problem is usually the lengths in the
header. Some software just complains when the lengths are wrong. Some
tries
Olle E. Johansson wrote:
LQ (Asterisk) wrote:
Hi guys,
I was reading that Steve Underwood is working on Asterisk R2 signalling
support, and has the 95% of the work done.
What is R2? I'm curious.
Half of R2D2, of course.
Its also a stupid clunky multi-tone based telephone signaling system
Eric Wieling wrote:
How does Grandstream become patent indemnified for their hardware? I
would assume they did not pay for a license for G723,1 and G729 directly
to the patent holding company. Maybe they did. I always assumed the
indemnification came with a DSP that implemented the codec.
Hi David,
David Liu wrote:
Hi there,
Anyone had any success deploying Asterisk with a T100P or T400P card
in Hong Kong? To my understanding, Hong Kong carriers only provide
IDA-P or IDA-M lines. I am looking to use IDA-P. Is this possible
with the card?
I know Cisco 2651XMV with a VIC
Martin Pycko wrote:
You have to contact www.openss7.org. The site may look dead but they
sell ss7 together with asterisk.
Yes and no. The sell access to the SS7 CVS. It does not work with
Asterisk. There is a project page about OpenSS7 - Asterisk integration,
but it is a project that never
Jan Czmok wrote:
Michael Devenijn ([EMAIL PROTECTED]) wrote:
Jan,
Where can we get any technical documentation about sccp protocol i've searched with google and at cisco but i don't find anything useful ...
The only useful resource is imagination :-)
Skinny is a Protocol developed by
the developer of libr2, Steve Underwood to comment on
this. its his code he knows best. Please comment
I keep commenting, and nobody seems to listen. libr2 is a half
implemented useless piece of rubbish. The real working R2 is not
available from me just yet.
Regards,
Steve
Hi Don,
A large number of GSM phones and PDAs now have bluetooth. It looks
likely that through 2004 the majority of GSM phones anywhere above entry
level will have Bluetooth. My guess is that this will collapse in 2005,
and bluetooth will be dead soon after. In the meantime, I don't seem
many
Christopher Lee wrote:
Hi Steve,
Interesting... I'm not sure! My copy of the original indications.conf had
400+17, and looking at the wiki it's the same there also
http://www.voip-info.org/tiki-index.php?page=Asterisk%20indications%20defaul
t
I tested 400*17 and it made a difference, but I still
PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Steve Underwood
Sent: Sunday, 25 January 2004 4:17 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk Indications
The correct tone is 400*17 (383 + 417) according to the ITU specs.
Actually, nothing would use a 17Hz tone
Christopher Lee wrote:
On Sun, 25 Jan 2004, Steve Underwood wrote:
Actually, nothing would use a 17Hz tone - it doesn't pass through a
300-3400Hz channel very well :-)
It's not a 17Hz tone. Australian (and others) tones are single-frequency
tones that are amplitude-modulated
Hi all,
I am interested in interfacing a GSM modem to *. I've seen a few
comments about doing this, but I'm not clear whether people have
actually made it work. I've used GSM modems for various data jobs,
mostly high volume SMS (no, not nasty marketing stuff - high volume
solicited SMS :-) )
Don Feuer wrote:
Hi Everybody,
In regards to what I see here, this looks like a whole .com flash back. I
started a phone company that went belly up (CentreCom, the first Unified
Communications company) because of customer service issues, lack of on-line
information, and a lack of caring for the
Alessio Focardi wrote:
Hello Jeremy,
Anyone can help me starting the card ?
JM List it on http://www.ebay.com/ and take the proceeds and purchase a
JM Digium E100P card.
It has been my first tought but guess what ? E100P is not CE
certified and I'm fearing legal problems
Steven Critchfield wrote:
On Mon, 2004-02-16 at 07:39, Jean-Marc V. Liotier wrote:
When dialing out, will a call be established significantly faster by an
ISDN adapter such as an Eicon Diva server compared to an analogical FXO
such as Digium's X100P ?
Analog, nothing logical there.
ISDN
Iain Stevenson wrote:
The problem with the Ofcom consultation as I see it is that it seems
to be regressive wrt to the position now being taken by the FCC.
There are probably not many more than 250,000 VoB users worldwide so
now is not the time to impose significant market constraints.
Why do
Hi Brian,
DTMF from a cell phone is rather different from DTMF from a land line,
but not in any way that usually makes a difference. The cell phone
codecs cannot carry DTMF properly, so the phone tells the base station,
by a message, that a particular key is being pressed. The base station
Hi Michael,
Michael Welter wrote:
I live at 8000' in the Rockies. We have lots of woodpeckers--they
especially love to drill 4 holes in the north side of my house.
They also like to drill on the arial telephone cables. Water then
gets into the cable and causes a partial grounding on the
Chris Albertson wrote:
Get a spectrum analizer.
Software will do it. Record the humming connetion to a file
and then run it through software that plots a power spectrum.
THere is plent of good open source software. Even some audio
file ditors have this feature. You should be able to see the
Greg Hill wrote:
My first thought was an RC filter, too. But I'd suggest that 500 Hz is too
high a cutoff, because a note like a middle C is 256 Hz. I don't think
it's uncommon for a voice (especially a male voice) to be in that range
frequently. Although (in English, at least) vowels generally
Chris Albertson wrote:
--- Steve Underwood [EMAIL PROTECTED] wrote:
A power spectrum plot will tell him he has a 60Hz hum. I think he
already knows that. I think he can definitely consider solutions
without
following your suggestion. :-)
No, It's not a 60Hz hum. Yes, 60Hz is getting
The left hand pedal on a church organ is around 16Hz. Below that things
don't really sound like tones any more.
Regards,
Steve
Matt Lawson wrote:
The low B string on a 5-string bass guitar is approx. 31 Hz
A power spectrum plot will tell him he has a 60Hz hum. I think he
already knows
Rich Adamson wrote:
Notching may not be that effective, as it will not deal with the
harmonics. The analogue to digital converter should already be
filtering below 300Hz, so you probably have quite a lot of hum if it
300Hz is pretty high to filter out... it's still well within the rage
Jose Quinteiro wrote:
I live at sea level, and have never seen a woodpecker going at any
telco equipment, but have a 60Hz hum on my POTS line through my Adtran
750.
It goes away if I pick up the telephone I have cross-connected on the
same line. Could it be the same problem (i.e., tip-ring
WipeOut wrote:
This is an interesting statement in the press release..
SIPxchange, the industrys first open source based enterprise
communications suite, is grounded in the concept that a community of
ideas provides a more fertile ground for innovation, progress and
product development.
I
Dave Cotton wrote:
Just done a cvs checkout on 2 different machines 1 running Mandrake 9.2
with a 2.4.25 kernel the other Mandrake 10.0 and 2.6.3
the most 10.0 fails at this point
chan_zap.c: In function `handle_init_r2_event':
chan_zap.c:4773: error: too few arguments to function `zt_new'
Costa Tsaousis wrote:
Also I would turn off Hyperthreading (in the bios). It
may cause problems.
What problems? Are these digium H/W specific, asterisk specific or
generaly Linux problems?
I don't know if The HT problems are generic, or something quirky in the
Zaptel drivers. However, if
Amaury Jacquot wrote:
Daniel Bichara wrote:
Hi Alex,
Alex G Robertson wrote:
Hi all,
I would like to have some information about your TE410p and TE405p
cards compatibility with telephony protocols adopted in Brazil.
- When in E1 mode, does it support R2 DIGITAL MFC 5C ?
You need a R2
If you want a freely usable implementation of SRTP look at
srtp.sourceforge.net.
Regards,
Steve
John Todd wrote:
I have found few VoIP clients that support encryption. The only one
that comes to mind is the Zultys devices (they have a softphone and a
hardphone that support SRTP.) I spoke
Robert Boardman wrote:
Hi
Just one question
do any of the Digium T1/E1 cards do DPNSS signaling?
Robb
Just one answer. No. :-(
Steve
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To
Jason Penton wrote:
Hi all
Does anyone know where I can get hold of the German 1TR6 ISDN signalling
protocol specification.
Thanks
Jason
Is that still used? I thought they were 100% CTR4 these days.
Regards,
Steve
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Asterisk-Users mailing list
Scott Stingel wrote:
If you'll be running commercial apps, I would recommend that you do a lot of
testing, especially load testing, with the types of applications you'll be
running. Dialogic boards, although incredibly expensive, do have lots of
horsepower built in for the purposes of encoding
hank smith wrote:
hello I am just curious if there is any windows alternitives to Asterisk?
can I also use them with free world dialup?
thanks
hank
No, but maybe you could port Asterisk to Windows. No, that's not a joke.
The Zaptel drivers might be tough, but Asterisk's VoIP features would
Michael Shuler wrote:
G.729 is just about everywhere. A lot of boxes use G.723 (and/or G.726) too
but G.729 ends up with about the same quality but at a much lower bit rate.
That's wrong. G.723 has the lowest bit rate amongst those codecs.
Most inexpensive hard phones don't use G.723 because
wangji wrote:
That's what Intel want to do, too. They guys have released a hardware
emulation software works like an four channel IP board. And they want to use
only interface board + host CPU instead of Dialogic products, so that they
needn't use DSP etc.
Have you actually tried playing
Hi all,
It seems this week's release of spandsp fixed the major problems in the
previous release, but still people have had a lot of trouble. Working
with some of those who tried the software and gave me good feedback, I
have identified some apparently common bugs in fax machines, and I have
Hi,
I have investigated some more fax machines that did not work with
spandsp, and made it more tolerant.
ftp://ftp.opencall.org/pub/spandsp/spandsp-0.0.1c.tar.gz is the result.
From what I have seen in today's investigations, I think this one will
work with considerably more quirky fax
Hi,
I have received more excellent problem report information, and I have
resolved a number of issues affecting my soft FAX machine when working
with various models of real FAX machine. The code now seems to be
working with a much greater range of fax machines. A problem affecting
the
Hi all,
If you have had trouble with multiple concurrent channels running
app_rxfax or ap_txfax, where was a silly bug. Updated versions are
available at ftp://ftp.opencall.org/pub/spandsp
The latest spandsp-0.0.1f seems to working for quite a lot of people. I
guess there will still be plenty
Tilghman Lesher wrote:
On Sunday 21 March 2004 07:10, Steve Underwood wrote:
I have received more excellent problem report information, and I
have resolved a number of issues affecting my soft FAX machine when
working with various models of real FAX machine. The code now seems
to be working
, Line 1744
(ast_pbx_run): Extension 8663222818, priority 1 returned normally even
though call was hung up
Thank you.
Alex Zarubin
Webley Systems
-Original Message-
From: Steve Underwood [mailto:[EMAIL PROTECTED]
Sent: Monday, March 22, 2004 8:14 AM
To: [EMAIL PROTECTED]
Subject
I think these products were bought by Aastra from Nortel. See:
http://www.aastra.com/corporate/Press/2000/pr012500-1_N.html
Regards,
Steve
Brian Johnson wrote:
Sure look the same to me
http://www.aastra.com/Products/archive/phones/Vista/can/vistaarchive.html
-Original Message-
From:
with it, using libPRI and not using
Asterisk.
The differences between
Steve Underwood writes:
[EMAIL PROTECTED] wrote:
Has any one used it?
Yes!
Regards,
Steve
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Hi John,
That is correct. Note those are VCXO crystals, and not ordinary ones.
They need to have the right pullable characteristics to get reliable
results. If you downlod the info from Dallas for the framer chip they
give some recommendations for off the shelf crystals which will work.
Regards,
[EMAIL PROTECTED] wrote:
Hi,Steve,
My ISA E1 card does not work yet. I would like to know if there are
other difference bettween then two kind of ISA card?
The crystals, the framer chips and the jumpers are the only differences.
What are the symptons of it not working?
Regards,
Steve
Claudio Aznar wrote:
Hello,
I'm testing the E400P with PRI signaling and work fine, but I want to test it
with R2 signaling.
My question are:
1. Can the E400P work with R2 signaling ?
2. If the firs question is yes, HOWTO?
Thank in advance
Peter
Is it available?
. How are working in this project ?
Regards,
Peter
On Tue, 11 Mar 2003 21:43:46 +0800
Steve Underwood [EMAIL PROTECTED] wrote:
Claudio Aznar wrote:
Hello,
I'm testing the E400P with PRI signaling and work fine, but I want to test it
with R2 signaling.
My question
Michiel Betel wrote:
Thats what I specifically asked... Should the whole system be approved (eg.
Computer, cards, software) or just the components. The answer I got back
from two agencies was that hardware approval under RTTE should be
sufficient. Same as ISDN BRI card manufacturers do...
There
Unless things have simplified since I was last involved in European
approvals (which is quite a long time) things are worse than that. If
your factory has not previously produced approved telecoms products, you
probably need to pay for a factory inspection; each new protocol you
want to
d hinton wrote:
FIRST LET ME STATE AGAIN I'M NOT A TROLL! my small company would like to
support the asterisk effort, but can't or won't pay a $800 markup. now,
please go to the http://zapatatelephony.org/ website and do some reading!
the pci card was their last card released, BY THEM not digium,
d hinton wrote:
Also, digium if i was you i'd destroy all evidence that i sold these cards
before your cert date, as they could fine you up to $250 per day (about $100
Grand per year), from the first date they can prove you sold them. that
would kill most budgets.
My understanding is that CC
Steven Critchfield wrote:
On Thu, 2003-05-29 at 15:46, Steven Critchfield wrote:
On Thu, 2003-05-29 at 15:06, Charles E. Youse wrote:
On 29 May 2003, Steven Critchfield wrote:
On Thu, 2003-05-29 at 10:44, Charles E. Youse wrote:
B8ZS is required for PRI. It's a digital
Mark Spencer wrote:
You could, of course, contribute the changes needed to make libpri do
Q.SIG, and everyone will benefit. :-)
What all is required for Q.SIG?
Another bunch of Q.931 type messages, basically. It builds on Q.931 to
add PBX (i.e. private network) specific features.
Regards,
Don Pobanz wrote:
When I was looking at timing before this is the conclusion that I have
come to.
The T400P card has an internal clock that all four T1s of that card
will be timed off of. This internal clock can be free running (not
referenced to any other clock) or reference to another clock
Jorge wrote:
Hi,
I have an E100P card.
My zaptel.conf is:
span=1,0,0,cas,ami,crc4
bchan=1-2
dchan=3
loadzone = us
defaultzone=us
after execute ztcfg red led becomes blinking, why ?
Anybody known the led's table of thruth ?
That looks like a pretty wacky configuration. ami? cas with a
Hi,
The D41/E does not support any sort of duplex audio path operation. That
seems a major limitation with Asterisk. What functionality can it
actuakky support with Asterisk?
Regards,
Steve
Matthew John Darnell wrote:
It should work, but there is a fee of $30 per channel for the software.
Rainer Jochem wrote:
We plan to have certification on the new TE410P board by the end of
summer.
TE410P?
*getting curious*
What kind of board will this be?
4 x T1/E1 ports, bus mastering. One board does T1 or E1 under software
control. I don't know if you can mix E1 and T1 modes on
It is normal. What you see depends on which version of various things
are on your system. The tor2 driver spends a lot of time in the
interrupt service routine (about 60% of the time on the 700MHz Athlon I
use). Whether the interrupt service times shows up as system usage, or
falls down a hole
Jan Rychter wrote:
Does G.729 provide better voice quality than GSM?
(a question for people who have tried both)
It depends. The bit rate of G.729 is a lot lower, so it starts with a
disadvantage. To overcome that, they made it a lot more complex and
tuned to the human voice. The result is
Matthew John Darnell wrote:
Why hasn't someone found 50 people who sound alike, put them in sound
studios and record the 10,000 most commonly used words. You would all
differnent forms of the 1,000 most words, i.e. leading, trailing, question
etc.
You can synthesize the other 0.05% when you run
Jeff Noxon wrote:
Many of you are familiar with how lousy Festival sounds.
ATT has a product, NaturalVoices, that sounds much better. There are
male female voice fonts for US/UK/Indian English, French, Spanish,
and German.
I am considering offering a linux-based text-to-speech engine based on
Moshe Yudkowsky wrote:
At 10:11 2003-07-16 -0700, Chris Albertson wrote:
SNIP
if you want a synthetic voice to sound
natural you will have to tell the software the _intent_ of the words
not just the words. You would need a markup language for that
emph I /emph said quotequestionword yes
LQ (Asterisk) wrote:
Dear fellows,
I need to use Asterisk with an E1 card with CAS R2 signalling for Argentina.
I know that the E100P don't support it right now.
Correct
Does anybody developing R2 drivers?
Yes.
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John Todd wrote:
LQ (Asterisk) wrote:
Dear fellows,
I need to use Asterisk with an E1 card with CAS R2 signalling for
Argentina.
I know that the E100P don't support it right now.
Correct
Does anybody developing R2 drivers?
Yes.
Interestingly terse reply; perhaps you can be more specific?
Hi Joe,
Most auto-dialers will accept commas in the dial string, and insert
delays where they occur. Will that work for you? Its normally used to
insert a delay after a 9 on a PBX, to get a stable outside line before
further dialing.
Regards,
Steve
Joe Antkowiak wrote:
Hi,
I am using a
David Boreham wrote:
P.S. Please do not answer again that this setup cannot work. In this
moment
I cannot accept such an answer.
Your e-mail made me chuckle. When I worked at Octel/Lucent
in the mid-90's we were constantly sniped at for trying to make
a voicemail system which ran on
Scott Lambert wrote:
On Sat, Jul 19, 2003 at 01:01:52AM +0800, Steve Underwood wrote:
That looks a bit like this one:
http://www.planet.com.tw/product/product_intro.php?menu_id=3
rather expensive to me. These things have less DSP and compute to do
than an ADSL modem, and should cost no more
OK Funny guy,
Mark Spencer wrote:
I put a TE410P card in a machine (a Tyan 2665 with 2x2.4GHz Xeons). A
red flashing light circles around the 4 RJ48C sockets. I load the
wct4xxp driver, and the flashing light stops. Whether I connect an E1
signal or not, no lights are shown, and no alarms are
Q: What's the difference between Asterisk and a softswitch
A: About $100,000
Soft switch - Hard to afford!
Regards,
Steve
Bruce Ferrell wrote:
I've been working in the VoIP industry for just a bit over a year
now... Mostly taking care of the underlying systems. I've now reached
the point
Hi Matthew,
That argument doesn't seem to work. I don't hear many complaints here
about the cost of the VoiceAge codec. It's the clunkiness of the
protection scheme people don't like. It's only the protection scheme
that seems to be making people want to dump the VoiceAge code.
Remember how
Eric Wieling wrote:
On Tue, 2003-08-12 at 15:37, Mark Spencer wrote:
Couldn't agree more. The G.729 codec is so unDigium-like... don't buy
it is my recommendation.
I don't think anybody buys G.729 just to have it. They buy it because
they *have* to have it. And we sell it because they
Scott Stingel wrote:
Hi all-
This question is for those familiar with EuroISDN setup.
I have a customer in Europe where I'm going to install an asterisk based
system with 4 E1's. The customer will configure them all in one large hunt
group.
My question is about the E1 channel configuration. I
Steve Underwood wrote:
The ITU G.729 code is pretty much useless for real world use. It is
very slow. It gets the right answers, but not by efficient means. All
the voice codec reference code I have seen is like this. The people
who develop these things *have* to write an efficient version
Dan wrote:
Hi Steve
Steve Underwood wrote:
06.10 isn't that great a codec,
though. I don't think it is used very much on the GSM networks these
days. Most of the time they use the enhanced full rate (EFR) or half
rate codecs.
What do you mean by isn't a great codec?
06.10 should
Hi Dan,
Dan wrote:
- Original Message -
From: Steve Underwood [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, August 13, 2003 9:49 AM
Subject: Re: [Asterisk-Users] Open G.729A codec
Steve Underwood wrote:
After writing this I got curious about how fast/slow the ITU
[EMAIL PROTECTED] wrote:
Hi all,
Trying to get fax reception going using a voicetronix openline4 card,
however, there are two issues (as far as I can see)
(1) Currently the voicetronix world (cards, firmware, drivers, channel
prog etc.) does not do fax tone detection. I have spoken to the
Hi,
Bob Knight wrote:
My thoughts on a DS3 * box:
Forget PCI. Forget x86.
There are very good bsd and linux ports for the powerpc.
There are ppc's with very good TDM interfaces.
All these framers and dsps speak TDM. Very simple clean design.
If you do not want to build any hardware, you can
John Todd wrote:
[...]
There have been discussions here on this list already on the
availability of boards like SBEI's channelized DS-3 card (they've been
a reasonably approachable vendor.)
Do they make such a thing? The DS3 cards on their site appear to be HDLC
data only.
Regards,
Steve
Manuel Marin Garcia wrote:
I compiled libtiff version 3.6.1 and spandsp and spandsp version k. When
trying to load asterisk I get the folloein error:
The instructions tell you *not* to use libtiff 3.6.1
Jun 7 10:15:03 WARNING[16384]: loader.c:408 load_modules: Loading
module app_dtmftotext.so
Chris Bond wrote:
Are they any issues still with hyperthreading processors, I've read and been
told by a few people to make sure its disabled in bios if I want to use * on
a hyperthreading machine.
A lot of people report no problems with HT turned on, but you have to
look at these reports
Hi Iain,
Your response seems to indicate that you don't know what HylaFAX and
spandsp actually do :-)
Regards,
Steve
Iain Stevenson wrote:
... might as well use hylafax.
Iain
--On Monday, June 7, 2004 2:15 pm +0100 Matt [EMAIL PROTECTED] wrote:
Hi all.
I'm looking to set up a fax via email
Kevin P. Fleming wrote:
Iain Stevenson wrote:
... might as well use hylafax.
Yes, well, that requires using modems and having Asterisk send the
audio back in/out as analog. It would be really fantastic if someone
could come up with an app for Asterisk that emulated a Class 1 FAX
modem and
Achilles Bochoris wrote:
Hello,
I understand these licensing issues very well. I don't reside in the
US, so I assume that there is no problem, especially for
testing/development, and not commercial use.
What I was asking however, is whether there is an alternative G.723.1
library which
Hi,
hskim wrote:
I heard that asterisk support r2 signaling.
I'm try to test r2 using e100p.
How should I configure zaptel.conf, zapata.conf?
And if I want to modify source for customization, where should I start?
Thanks.
Asterisk does not support R2 right now. You will find some R2 code
Kevin P. Fleming wrote:
Steve Underwood wrote:
spandsp doesn't try to reimplement all of HylaFAX. It reimplements
only one piece - the T.4/T.30 code. I have a half implemented
spandsp as class 1 fax modem which I put aside. People are using
spandsp happily for things like fax to e-mail
Scott Nelson wrote:
My office is investigating using an Asterisk PBX and also going to a VOIP
provider for our main phone connections, but one of the tricky things is that
we need to have outbound and inbound modem calls (fax too).
I see a lot of talk about faxes but no mention of modems on
Darren Nickerson wrote:
Steve,
HylaFAX supports 1D MH, 2D MR, and 2D MMR.
The last time I looked (a few months ago) it supported those file
formats, but only supported 1D transfers on the wire.
ECM is new in HylaFAX, but already seems more robust than the implementation
one finds in most
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