Michael Welter wrote:
Can someone explain where we are with spandsp? Is it ready for a
production environment? How much will one fax using spandsp load the
processor on an * system?
Thanks,
Mike
Some people have been using spandsp-0.0.1k in production systems for
months. The main problems hav
Danny Froberg wrote:
Hi folks,
Working on getting AlarmReceiver to work on newer SIA protocols and
have some thoughts if anyone has used i.e. t38modem to receive the
short bursts of data that an alarm communicator sends inside Asterisk.
This is a lot simpler than i.e. receiving a fax, so maybe s
Tom Neville wrote:
I've been running ssh tunnels for a couple of years now. For years,
they've worked well. However, now that I've got asterisk up I do
notice problems. Biggest indication of this is if I'm on a call and
run a program in another window that scrolls and scrolls call quality
dr
See http://www.opencall.org/faq/x26.html
Rodger Lewis wrote:
Using 10/12/04 cvs of asterisk and spandsp.0.0.2pre4
After changing line 86 in app_rxfax for new callerid info i got a clean
compile.
Using tiff-v3.5.7 straight from the tiff site and compiled manually no
packages.
I am getting half pages
Vladyslav wrote:
Hi All.
How to receive multiple pages with rxfax ?
Here is what I have:
exten => 10,1,Setvar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten => 10,2,Setvar([EMAIL PROTECTED])
exten => 10,3,rxfax(${FAXFILE})
exten => 10,4,system(/usr/local/sbin/mailfax ${FAXFILE} ${EMAILADDR}
Darren Sessions wrote:
What is the rational for only supporting 32kbps G726 and not 16kbps?
Thanks,
G.726 32K is widely used. The other bit rates are not. That seems one
sensible rationale for only supporting 32K.
Regards,
Steve
___
Asterisk-Users maili
Graham Turner wrote:
have posted a while ago on issues of receiving faxes by an Asterisk host
using an x100p fxo interface attached to BT pstn
the asterisk installation is the cvs download as of 23/09/04
is anyone able to confirm that the rxfax / txfax application that seems to
be 'bundled' in thec
Michael Bielicki wrote:
find someone to host it in India or serbia and you can safely ignore it :)
On Sat, 25 Sep 2004 10:22:52 +0200 (CEST), Peter Svensson
<[EMAIL PROTECTED]> wrote:
On Sat, 25 Sep 2004, Steve Underwood wrote:
On Sat, 25 Sep 2004, Steve Underwood wrote:
I am not a
Well maybe you should be a user. I offer much less than *, at only a
much greater cost :-)
I think this is a bit like advertising Windows XP on the Linux kernel
mailing list :-)
Regards,
Steve
Steve Totaro wrote:
I am not an OnDo user. Please do not spam me.
- Original Message -
From:
Peter Svensson wrote:
On Sat, 25 Sep 2004, Steve Underwood wrote:
I wouldn't do that, if I were you. Distributing the source code for
educational and evaluation purposes won't get anyone into trouble with
the patent issues. I think (not sure) that Intel's copyright on the cod
Eric Wieling wrote:
On Fri, 2004-09-24 at 20:33, Steve Underwood wrote:
It is very difficult to be legally correct with this. The IP holders
don't have simple programs for selling licences in small quantities. If
you buy licences from Digium, they deal with the IP issues on a larger
v
Arkadi Shishlov wrote:
I expropriated the right to rip Daniel's disclamer for use in my
email too..
DISCLAIMER:
You might have to pay royalty fees to the G.729 patent holders for using
their algorithm.
For easier testing I prepared codec_g729.so binaries and associated
libraries and put them on the
Danny Zak wrote:
Hello TELUX,
could anybody post something more about being legaly correct using
this codec and the corresponding "royalty's".
It is very difficult to be legally correct with this. The IP holders
don't have simple programs for selling licences in small quantities. If
you buy li
better tested. If you have the equipment ready to try MFC/R2 please
tell me how you get on.
Regards,
Steve
Steve Underwood wrote:
Hi all,
I have begun the release of my MFC/R2 protocol software. At
http://www.opencall.org/installing-mfcr2.html there are instructions
for installing what I have
Some switches are fussy about you getting the NPI and TON (sometimes
jointly known as the dial plan) right. That is usually the cause of the
problem you see.
Regards,
Steve
Paul Oster wrote:
I'm in the process of turning up a PRI in one of my markets and have
run into a problem I have never seen
Hi all,
I have begun the release of my MFC/R2 protocol software. At
http://www.opencall.org/installing-mfcr2.html there are instructions for
installing what I have released so far. This is the MFC/R2 protocol
software, and a test program. The software to interface Asterisk to the
MFC/R2 code wi
Michael Bielicki wrote:
On Mon, 20 Sep 2004 08:43:35 +0100, Tim Robinson <[EMAIL PROTECTED]> wrote:
Guys - R2 SIGNALLING IS NOT SUPPORTED IN ASTERISK. How many times does
this topic come up on the list? Steve Underwood posted something like
this last week! I will save him the trouble
[EMAIL PROTECTED] wrote:
Thanks, I applied that patch and get an error on modules load after
rebuilding *:
[app_rxfax.so]Sep 20 10:08:59 WARNING[-1084751200]: loader.c:248
ast_load_resource: libspandsp.so.0: cannot open shared object file: No
such file or directory
Sep 20 10:08:59 WARNING[-1084
Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Underwood
Sent: September 20, 2004 2:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] English vs American voice files
[EMAIL PROTECTED] wrote:
On 20 Sep 20
administrator tootai wrote:
Graham Turner a écrit :
I am attempting installation of spandsp on to an Asterisk
installation on
Linux RH9
the distribution i am using is that are URL http://ftp2.tootai.net - the
README for which i have followed verbatim -
It's not a special distribution, it's th
Sam Njenga wrote:
I have a digium D300-E1 card and the R2 link ready. Actually it is
already connected to the card and all alarms cleared. Am not in
Argentina but the R2 signaling is configured with Argentina signaling.
It works very well when I connect a Cisco 5350 and configure it to
Argentin
[EMAIL PROTECTED] wrote:
On 20 Sep 2004 at 12:38, Andreas Sikkema wrote:
[EMAIL PROTECTED] wrote:
Initially we recorded using 16 bit/8K sampling on the basis
that this is what is required by Asterisk but that was
really terrible. So we're sampling at higher rates on
the basis that we can
Bill Seddon wrote:
My wife has been recording the text published on the wiki. A couple of
questions for you:
1) One of the recordings says "please enter the full 10 digit number
starting with the area code". Any opinions on whether this should be
changed for the UK and, if so, to what?
2) The rec
This is complete rubbish. The default for that code is China R1,
although some measure of partial support for Argentina in in there too.
I wrote that code, and it is useless. It works as well for incoming
calls as it does for outgoing - its useless. When I did a proper
implementation of R2 sign
Put another way, if BT call it square, its almost certain nobody else in
the UK does :-)
Regards,
Steve
David Davies wrote:
Most Pbx's I have worked with use hash in the uk.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips
Sent: 17 September
TC wrote:
Roger
I'm currently beta testing the TE410P with SS7 together with
a partner, who will present SS7 support for asterisk is some
weeks, maybe some days.
GULP :)
Is this GPL effort ?, and expected to be a standard part of asterisk cvs ..?
ps. never fails to amaze what going on the b
Bob Knight wrote:
Steve Underwood wrote:
Arinze Izukanne wrote:
Hi Guys,
Does anyone know of E3 PCI cards that work with
Asterisk?
Arinze
No, but if you find an E3 PCI card with nice Linux support there
might be people interested in helping to get it working with *.
SBE (side band engineering
Even with the robbed bit thing you get 62666.7 bits/s, since it only
steals the LSB every 6 samples. :-)
Regards,
Steve
Marcelo Pacheco wrote:
A T1 is 24 64000bps channels.
The 56000bps thing is when robbed bit signalling is used, it steals bits from
each voice channel for call signalling, while
Adam Goryachev wrote:
On Wed, 2004-09-15 at 04:29, Lee Howard wrote:
On 2004.09.14 11:10 Marty Mastera wrote:
2)Packet loss, etc...makes faxing over the internet unreliable
I'm not sold on this theory yet. I don't think that it's so much a
matter of packet loss (this shouldn't
Arinze Izukanne wrote:
Hi Guys,
Does anyone know of E3 PCI cards that work with
Asterisk?
Arinze
No, but if you find an E3 PCI card with nice Linux support there might
be people interested in helping to get it working with *.
Regards,
Steve
___
Aster
Rob Fugina wrote:
I've recently started playing with the RxFax application on my
Asterisk box. I've had success, mostly, but I've had some failures,
too...
The most recent failure is specific to receiving from a particular fax
machine -- a "Canon Laser Class 9000S". The TIF images received are
re
Andreas Sikkema wrote:
Hi,
We've got endpoints and gateways who have T.38 fax support. We
now use SER and Asterisk to do our routing and other
functionality, but fax doesn't seem to work. Asterisk complains
like this:
Sep 9 09:25:45 WARNING[467828746]: RTP Read too short
Sep 9 09:25:45 WARNING
Chris Lee wrote:
Steve Underwood wrote:
Chris Lee wrote:
I am looking at building an IVR product with a few interesting
features and need some more information about how asterisk and VoIP
work and what I can get from them.
As far as I can tell when I use ISDN/GSM telephone networks the DTMF
Chris Lee wrote:
I am looking at building an IVR product with a few interesting
features and need some more information about how asterisk and VoIP
work and what I can get from them.
As far as I can tell when I use ISDN/GSM telephone networks the DTMF
information travels as data representing 's
Eric Jacksch wrote:
Are there any codecs that are particularly good for fax traffic? Any to avoid?
---
Eric Jacksch
[EMAIL PROTECTED]
See http://www.opencall.org/faq
Regards,
Steve
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.
Hi,
T.38 should be completely insensitive to the number of hops. That is its
whole reason for existing. It sounds like these units are not using T.38.
Regards,
Steve
Leo Ann Boon wrote:
Try welltech 3502 (2-port) or 3504A (4-port). beware it only works if
your 2 endpoints are not too many hops a
Chris Shaw wrote:
- Channel Support:
IAX2 in asterisk
IAX2 in libiax2
Other IP channels in asterisk (RTP-based ones, I guess are all that is
left).
CNG/VAD and DTX in SIP is a must if * is to be taken seriously as a complete
solution... As much as we all hate it's complexity and wish that
[EMAIL PROTECTED] wrote:
On 30 Aug 2004 at 0:26, Steve Underwood wrote:
[EMAIL PROTECTED] wrote:
Why doesn't asterisk clock to the 1000 interrupts per second instead
of the incoming audio? Were there no interrupts available when it
started? Even if you had no card you could us
Joseph Shi wrote:
Does anyone know if there are any reseller for the book "VoIP
Telephony with Asterisk" in Hong Kong/Asia region? I'm interested in
purchasing the book but the shipping charge to Hong Kong is expensive.
Thanks.
Joseph
Just wait for the simplified Chinese version to appear in S
[EMAIL PROTECTED] wrote:
Why doesn't asterisk clock to the 1000 interrupts per second instead
of the incoming audio? Were there no interrupts available when it
started? Even if you had no card you could use the ztdummy module
and even though that might be off by a bit, surely it'd sound better
Hi,
The following part of your log:
Fast carrier up
Ouch ... error while writing audio data: : Broken pipe
Ouch ... error while writing audio data: : Broken pipe
Floating point exception
would seem to indicate some problem with the libaudiofile on your
machine. I guess libaudiofile must be install
LOG_FAX_AUDIO
/*
* SpanDSP - a series of DSP components for telephony
*
* t30.c - ITU T.30 FAX transfer processing
*
* Written by Steve Underwood <[EMAIL PROTECTED]>
*
* Copyright (C) 2003 Steve Underwood
*
--
I think is right uncomment but i dont see ant audio log under /tmp, do you
th
Nana Yaw wrote:
Hi,
How did you find it in your locality?
Eg, I am in London and have a T Mobile and an O2 mobile.
I will check with them to see what they say first of all.
Regards
Leslie
Do you know a means to get a complete and honest answer from a telco? :-)
Regards,
Steve
___
Stopping in mid page is usually a timing problem. See the spandsp FAQ.
Regards,
Steve
Jean-François Rousseau wrote:
Hi , does anybody have successfully received a full fax with spandsp ? I
keep having only about a quarter of the page and then the other part is
garbage. Does anybody have any solutio
In most countries the legislation which licences the cellular operators
to use certain spectrum for certain types of communication also controls
who may provide publicly offered (and sometime privately offered)
interworking services. This may also apply to wireline services. It is
very country
EREDTIME=") in new stack
-- Executing Hangup("Zap/35-1", "") in new stack
I dont know how to debug more, you can give more help to trace the problem?
Thanks in advance.
Dimitri
On Wednesday 25 August 2004 11:34, Steve Underwood wrote:
Hi,
Several people have re
Hi,
Several people have reported problems sending faxes from spandsp-0.0.1k
to Canon FAX machines. A spandsp user had the same problem with another
make of FAX machine, and traced the problem to a bug in the file t30.c
of spandsp. Line 542 says s->t4.rx_file[0] where it should say
s->t4.tx_file
Hi,
Several people have reported problems sending faxes from spandsp-0.0.1k
to Canon FAX machines. A spandsp user had the same problem with another
make of FAX machine, and traced the problem to a bug in the file t30.c
of spandsp. Line 542 says s->t4.rx_file[0] where it should say
s->t4.tx_file
Hi Vladyslav,
Several people with these symtoms - crashing just as the reception of
the actual page starts - found they had other versions of libtiff on
their system, as well as 3.5.7. When the others were removed the problem
when away.
Regards,
Steve
Vladyslav wrote:
HI All.
I'm using tiff-v.3
Andres wrote:
[EMAIL PROTECTED] wrote:
On Tue, 3 Aug 2004, Steve Underwood wrote:
Eh? G.729 has no particular features to allow more effective packet
loss concealment. iLBC has, but at the cost of a substantially
higher bit rate. In fact G.711 is a little ahead of G.729 in the
regard, since
Adam Hart wrote:
Steve Underwood wrote:
Adam Hart wrote:
Daniel Niasoff wrote:
Is G729 more sensitive to packet loss or delays due to it’s higher
compression. If I’ve generally got the bandwidth available, am I
best sticking to ulaw.
G.729 has lost packet concealment, G.711 doesn't. G.711
Adam Hart wrote:
Daniel Niasoff wrote:
Hi Everyone,
Is G729 more sensitive to packet loss or delays due to it’s higher
compression. If I’ve generally got the bandwidth available, am I best
sticking to ulaw.
G.729 has lost packet concealment, G.711 doesn't. G.711 will sound
better otherwise i
Daniel Niasoff wrote:
Hi Everyone,
Is G729 more sensitive to packet loss or delays due to it’s higher
compression. If I’ve generally got the bandwidth available, am I best
sticking to ulaw.
Thanks
Daniel Niasoff
Neither G.729 or ulaw have any strong features to help with packet loss
concealment
just sent this to Steve Underwood, but then found a bunch of posts on the
mailing list about similar issues.. does anyone have the fix?
I'm running asterisk CVS-HEAD-06/28/04-18:13:13, spandsp 0.0.1k, libtif 3.5.7
one thing i just noticed is that calls come in with format '72' w
TED]>
Sent: Thursday, July 29, 2004 11:28 AM
Subject: Re: [Asterisk-Users] faxing
> What's wrong with
> ftp://ftp.opencall.org/pub/
>
> It says "Can't open data connection"
>
>
> On Thu, 2004-07-29 at 17:44, Steve Underwood wrote:
> > [EMAIL
Holger Schurig wrote:
This is phone and the ATA is available soon from
http://www.eezeephone.com priced at $75.00 each.
This is one more phone based on the PA168 chipset. I guess they're all
compatible with Asterisk.
I recently added the pages "Atron" AND "PA168" to the wiki.
How did you
[EMAIL PROTECTED] wrote:
What are your experiences with faxing through Asterisk to the PSTN?
We are using g.711u as a codec, and are originating/terminating with Broadvox as
well as through our own PSTN gateways.
We have had some luck with incoming faxes coming into our network from Broadvox
DIDs.
Hi David,
96 and up are dynamically allocated codecs, so more information is
needed to tell what the codec really is. On the general topic of FAXing
over VoIP did you look at http://www.opencall.org/faq ?
If your VoIP provider is half decent, they should be able to deliver FAX
to you by T.38
R
Carlos Hernandez wrote:
I use FC2, and I have found the following:
I did not try * with the distro's kernel. I had not started with my
testing with asterisk.
I upgraded to kernel-2.6.6-1.435.2.1
I then started my testing, and had troubles loading the zaptel modules
Someone in the IRC reccomended t
Leif Madsen wrote:
On Mon, 26 Jul 2004 13:58:04 -0700, Florin Andrei
<[EMAIL PROTECTED]> wrote:
On Mon, 2004-07-26 at 13:12, Leif Madsen wrote:
ztdummy works fine on FC2. I was able to get a TDM400P to work first
try.
Using the distro kernel, or the vanilla 2.6?
Distro kernel.
Duct tape applied to the mouth is cheaper, and is a more effective noise
reduction technique. These days people just go straight for the high
tech solution, and skip all the traditional technology, that often gives
superior performance. :-)
Steve
David Hickman wrote:
My wife used to pay for alo
Hi,
The SL1 was an old Northern Telecom PBX, from the late 1970s/early 1980s
- the precursor to the Meridian. I've never seen it refered to as a
protocol. Now, if you really means the Meridian Link CTI protocol, then
yep, I know about that. They charged a fortune ($25,000 I think) for a
copy of
bit123 wrote:
hi!
What's the libr2 status for Asterisk ? I've got R2 E1 delivered to my * box.
I have TE410P digium quad card with newest CVS.
How much % is completed with libr2 ?
what's completed ?
& What's missing ?
Thanks,
bit123.
libr2 gives you about 5% of a very bad R2 implementation. I wi
John Galt wrote:
could one at least in the case of the fxo/fxs cards just call out one
port and be looped back into the other, record the outgoing and
incomming call (one recording / port) then compare the phase
difference of the 2 recordings?
-Galt
That is probably the simplest way to achieve
Rich Adamson wrote:
On Fri, 2004-07-16 at 12:07 -0600, Rich Adamson wrote:
No echo on eMachine T2240 2.2ghz Celery, 360m RAM, with either tdm04b
or x100p running any Head cvs after June 23rd (totally stock install).
Wouldn't necessarily recommend this box for any commercial production
use, but
[EMAIL PROTECTED] wrote:
From the CLI and during a call I want to be able to:
*** Pulse the outgoing line and record at least 50 ms of the incoming line.
The pulse waveform must be specifiable as a series of amplitudes
for each 1/8000 sec time slot. It would be best of these values
Steve Underwood wrote:
Andrew Kohlsmith wrote:
On Saturday 10 July 2004 11:21, Rich Adamson wrote:
If you install a T1 card and an external T1 mux (with fxo cards), the
echo can function already exists within the mux and/or cards. Don't
really need 'another' external echo can
Andrew Kohlsmith wrote:
On Saturday 10 July 2004 11:21, Rich Adamson wrote:
If you install a T1 card and an external T1 mux (with fxo cards), the
echo can function already exists within the mux and/or cards. Don't
really need 'another' external echo can box unless you actually
purchased a T1 mux
Rich Adamson wrote:
[...]
If you install a T1 card and an external T1 mux (with fxo cards), the
echo can function already exists within the mux and/or cards. Don't
really need 'another' external echo can box unless you actually
purchased a T1 mux that didn't have echo can in the first place (and
t
The switches already support this. In most parts of the world an end
user trunk can only use a caller ID within their allocated blocks of
numbers. Attempts to use other caller IDs usually result in the call
being rejected. In some cases it results in the call completing, but the
receiver sees a
Andrew Kohlsmith wrote:
On Thursday 01 July 2004 01:19, Jay Milk wrote:
That would be a great alternative. For what it's worth, the phone is
based on a PA1688 single-chip VOIP terminal, which in turn contains a
50MHz 8051-compatible and a ADSP2181 DSP running at 33MHz. The Sound
interface is A
Joseph wrote:
Does anyone know of a device that will take an SS7 link and convert it
to a PRI?
It could be * - depending which version of * you have. :-)
Regards,
Steve
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/l
Kevin P. Fleming wrote:
Lee Howard wrote:
I stand corrected.
After a little bit of work with the fax application to adjust the
timings (increasing all of the pauses), all is well with V.17 also.
I assume you're using no compression (G711u) between the X100P and the
SPA-2000, then. Are you findin
Simon wrote:
Hello
I have contacted my line provider who is saying that in order to get my 0845
or 0870 number to id as the incoming number on a landline that i may call i
need the following.
User must provide - NPI set to E.163/E.164
User must provide - TON = "national or international
I have had
Florian Overkamp wrote:
Hi,
-Original Message-
VoiceXML support would be great, but I know of any active work on it.
openVXI seems to have spri=ung to life again recently, after years of
languishing. Perhaps it would form a sound base to get
VoiceXML up and
running in a reasonable t
Hi,
VoiceXML support would be great, but I know of any active work on it.
openVXI seems to have spri=ung to life again recently, after years of
languishing. Perhaps it would form a sound base to get VoiceXML up and
running in a reasonable time.
Regards,
Steve
Asterisk User wrote:
Hi All,
Do any
Lee Howard wrote:
I've never seen this kind of "flakiness" of libtiff cause any problems
for HylaFAX. As far as I'm aware, there has only been two instances
when libtiff caused HylaFAX any grief. The 3.6.1 release problem with
G3/G4 is a given. And then there was the 16-to-32 bit type change
Holger Schurig wrote:
Unless someone does something serious about the flakiness of libtiff, I
don't think either spandsp or Hylafax will ever be very stable. :-(
Delete the word "unless".
And then create a subdirectory spandsp/tiff where you put a libtiff into
it that actually works. Create t
e you join.
This was needed due to the spambots and the few abusive people.
bkw
- Original Message -
From: "Steve Underwood" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, June 19, 2004 10:40 AM
Subject: [Asterisk-Users] IRC
It seems the #asterisk c
It seems the #asterisk channel on IRC has become an exclusive club.
Suddenly it gives:
An access level of [5] is required for [INVITE] on #asterisk
What's up?
Regards,
Steve
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/ma
Lee Howard wrote:
Furthermore, even if you assumed that spandsp was as stable as
HylaFAX, there is a vast feature-set difference between them as far as
the faxing itself goes. Steve has already made it clear that he sees
no future in fax, and that he does not intend to bridge that
feature-set
Klaus-Peter Junghanns wrote:
Am Fr, 2004-06-18 um 17.53 schrieb Darren Nickerson:
You don't even need spandsp - fax is dead, remember? ;-)
Why do YOU sell hylafax servers then? ;)
best regards
Klaus
Working with the dead never stopped undertakers making a living :-)
Regards,
Steve
__
Andrew Kohlsmith wrote:
On Friday 18 June 2004 11:10, Klaus-Peter Junghanns wrote:
better send the EUR 10k (not $10k... :) ) to the author of spandDSP.
Nobody needs HylaFAX for receiving faxes. Converting a tiff to pdf and
storing it somewhere is not rocket science. ;)
Incorrect. I've bee
Klaus-Peter Junghanns wrote:
better send the EUR 10k (not $10k... :) ) to the author of spandDSP.
Nobody needs HylaFAX for receiving faxes. Converting a tiff to pdf and
storing it somewhere is not rocket science. ;)
best regards
Klaus
I'd just like to point out that this kind of thing *is* perf
Michael George wrote:
In the mailing list archives, I found a message that indicates that
the IAXy has the ulaw, alaw, and g726 codecs, but I cannot find
anything official on Digium's site about it. The Installation Manual
has an example iax.conf file that indicates the ulaw codec, so I know
t
Klaus-Peter Junghanns wrote:
TDMoIP is nothing else like IAX2 with trunking, i would say. And a
compression of 16/1 (payload bandwidth!) sounds like g723.1 to me.
16:1 means an avaerage of 4kbps per channel. It would have to be G.723.1
with optimistic silence compression to get that low. I gue
Hi Mike,
To get something like:
Coarse carrier frequency 1832.96 (4)
Training error 927.702492
Training failed (convergence failed)
something is horribly wrong. The carrier should be 1700Hz, not 1832.96Hz :-)
Do you have a codec mismatch, or are you using a codec other than u-law
or A-law? Sometim
Hi Patrick,
I can't tell much from this brief description. Send me a console log.
Regards,
Steve
Patrick J. Conroy wrote:
Hello All,
I have downloaded and installed spandsp and downloaded rxfax, etc and
rebuilt asterisk with app_rxfax. I have added the following to my
extensions.conf:
[macro-faxre
Kurt wrote:
Old managers will change its the LaLawyershat don't
change. Every dam law office that I been in has at
least one fax machine that is constantly printing
something out. But to say fax is dead is an
understatement.
AT&T said that about teletype service, you know 50 -
300 baud service, y
Lee Howard wrote:
On 2004.06.11 20:47 Steve Underwood wrote:
The last info I got from a large FAX server is about a year old. It
seems after several years of nothing much changing, FAX has suddenly
taken a step up - kind of sad it should improve now it is obsolete :-)
Fax was only partially
dkwok wrote:
I need clarification as to DID in T1 connection.
T1 provides 24 channels for voice/data. Do it assign each channel to
particular DID. Or you can have unlimited DID to share the 24 channel
as an example. ie. Outgoing/incoming traffic is not bound to
particular channel. Whatever is av
Darren Nickerson wrote:
The last time I checked on a big FAX server, only a few percent of the
calls used anything but basic 9600bps non-ECM operation. When I look in
the shops, hardly any of the FAX machines - other than the low selling
high end laser models - support anything fancy. If you are de
Randy Ackers wrote:
Tony Hoyle wrote:
Steve Underwood wrote:
I didn't say one patent covered all the world. I said the patents on
codecs exist all over the >>world. WIPO is simplifying this a bit,
but its still pretty expensive to get a patent everywhere. I >>know
of no coun
The reference code does not pack or unpack the bits. It needs additional
work to make a usable codec. This is true of most reference codec
implementations. The bit packing arrangements depend on the application
of the codec, so they are often not specified as part of the codec.
Regards,
Steve
V
Tony Hoyle wrote:
Steve Underwood wrote:
I didn't say one patent covered all the world. I said the patents on
codecs exist all over the world. WIPO is simplifying this a bit, but
its still pretty expensive to get a patent everywhere. I know of no
country where the key aspects of a codec c
Steve Kennedy wrote:
On Tue, Jun 08, 2004 at 07:06:22PM -0700, George Pajari wrote:
http://www.nwfusion.com/columnists/2004/0607faceoffyes.html
There are very valid arguments in the contra argument. If you have
existing equipment it's all about integration. Traditional telcos are
moving to
Holger Schurig wrote:
Codecs are patentable and patented worldwide.
I'm not a lawyer --- but patents are not valid world-wide. Some countries
have mutual patent agreements, other countries haven't. Some countries
permit patents on everything, some are more restrict.
I didn't say one paten
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 consu
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 this
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 lik
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