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
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
Nick Bachmann wrote:
Hi Michael,
Michael Welter wrote:
the codecs I use? Filter-out everything between, say, 55 and 65Hz?
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 pro
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 ha
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
Rich Adamson wrote:
300Hz is pretty high to filter out... it's still well within the rage of
voices. To compare, 300Hz is about a diatonic concert D.
POTS has been filtering out < 300HZ and > 3000HZ for years.
Only on long distance equipment, not on local plant. Ringing frequencies
of
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 cir
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
the
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
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 w
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
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
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 :-) ) .
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-modu
sounding.
-Original Message-
From: [EMAIL 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
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
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
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
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 Se
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 we
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 ca
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.
Mo
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 s
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 to
Rich Adamson wrote:
What are the practical effects with in-correct clock sync
-like to you hear odd buzzing, or dropped voice or gaps of audio ??
You may get gaps where frames are discarded, this will be across all timeslots
so an individual loss isnt a lot of data, you'll probably get awa
Rich Adamson wrote:
To complete this rather lengthy topic... what happens if you ignore all of
this and just slap a bunch of systems together with no regard to a master
sync source? The quality and stability of your network will likely not be
as good as what it could be. If your clocks (in each d
Stephen Davies wrote:
On Wed, 14 Jan 2004, TC wrote:
What are the practical effects with in-correct clock sync
-like to you hear odd buzzing, or dropped voice or gaps of audio ??
Old-fart anecdote about this - in the early 80s we had some 1200bps
modems that we used to connect to client s
TC wrote:
To complete this rather lengthy topic... what happens if you ignore all of
this and just slap a bunch of systems together with no regard to a master
sync source? The quality and stability of your network will likely not be
as good as what it could be. If your clocks (in each device) hap
ublic carrier, and therefore have no idea where to start.
Would someone (Steve Underwood ;-) )mind at least putting me on the right
track so I can address this issue?
Thanks in advance Steve
Jason
I don't know why this would fail. An ISDN card should be properly
synchrinised to the PSTN, and use
John Brown (CV) wrote:
It appears that zttool doesn't actually report T1 span
errors.
If I inject BPV's, crc errors, framing errors, etc into
a T1 span, the counters on zttool don't change.
It works OK for me with Tormenta 2 and TE410P boards. Both zttool and
the /proc/zaptel/x files seem to
Anton Tinchev wrote:
Steve Underwood wrote:
Anton Tinchev wrote:
Richard Grinnell wrote:
Dell - PowerEdge 400SC Server Under $300 with MIR
Intel® P®4 Processor at 2.8GHz, 512KB Cache, 800MHz
FSB
For those of you who aren`t familiar with the 400SC, this server is
an Intel i875P chipset based
Anton Tinchev wrote:
Richard Grinnell wrote:
Dell - PowerEdge 400SC Server Under $300 with MIR
Intel® P®4 Processor at 2.8GHz, 512KB Cache, 800MHz
FSB
For those of you who aren`t familiar with the 400SC, this server is
an Intel i875P chipset based server
with an 8x AGP slot. It is compatible wit
lling differences between countries ( I am
trying to run .ar at .br)?
Daniel
Steve Underwood wrote:
Hi Daniel,
You will find libr2 is only about 10% of an implementation, and a bad
one at that. I now have 95% of a good implementation, but its not yet
released.
Regards,
Steve
Daniel Bichara
John Todd wrote:
hurricane
tornado
You missed typhoon!
Regards,
Steve
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Hi Daniel,
You will find libr2 is only about 10% of an implementation, and a bad
one at that. I now have 95% of a good implementation, but its not yet
released.
Regards,
Steve
Daniel Bichara wrote:
Hi all,
I will start testing libr2 for brazilian R2. Any clue?
Daniel
__
Anton Tinchev wrote:
Just spended ~ hour googling - all boards are based on GC-XX or I750X
Chipsets - all for Xeons. There also some boards for Pentium 3.
Can someone point me to a board with 64Bit 3.3v PCI for ordinary P4
with 800Mhz FSB.
Thanks
Unless one has appeared in the last couple of wee
9's allows about 6
minutes downtime a year. That means 100% of all failures must have
automated failover, as manuals repair could never be achieved so fast.
Physical diversity if essential for that.
Regards,
Steve
Chris Albertson wrote:
--- Steve Underwood <[EMAIL PROTECTED]> wrote:
WipeOut wrote:
Granted five 9's is never easy but in a cluster of 10+ servers the
system should survive just about anything short of an act of God..
You do realise that is a real dumb statement, don't you? :-)
A cluster of 10 machines, each on a different site. Guarantees from the
power company
Patrick Wong wrote:
Hi all,
I just checked out that Asterisk which is a platform I am interested
of. I would like to install it to the Linux box for a trial. I have
some legacy Dialogic hardware on hand, don't know they will work with
Asterisk or not. For analog loop start interface I have
That status page tells you the porject has gone nowhere so far. What you
need is not a driver. It is a development project!
Regards,
Steve
Ray Burkholder wrote:
Current Status: http://www.openss7.org/asterix.html
Ray
Do I need a special Digium Card (E100-SS7) or use my E100P card and compile
And SRTP uses AES!
Steve
Brian West wrote:
I understand AES can do this.
bkw
On Thu, 25 Dec 2003, Steve Underwood wrote:
Steven Critchfield wrote:
On Wed, 2003-12-24 at 19:11, Mahoney, Matt wrote:
Hi,
Does asterisk support any kind of voice encryption?
Not
Steven Critchfield wrote:
On Wed, 2003-12-24 at 19:11, Mahoney, Matt wrote:
Hi,
Does asterisk support any kind of voice encryption?
Not right now. As I understand it, it is a problem with the fact that
each packet would have to be able to be decrypted even if packets in the
stream are los
Eduardo Goncalves wrote:
On Mon, 22 Dec 2003 15:48:37 -0600
Steven Critchfield <[EMAIL PROTECTED]> wrote:
asterix:~# modprobe tor2
Zapata Telephony Interface Registered on major 196
Detected Tormenta 2 Quad T1/PRI or E1/PRA at 0xfe121800/0xfe121000
irq 7 Did not get DONE signal. Short file mayb
Steve Totaro wrote:
Supports H323
http://www.viseon.com/prod/c_VisiFone.asp?id=133
So? Whilst there are still only a few VoIP audio phones available,
almost every computer related manufacturer in Asia has at least one
video phone model like this. There must be dozens of units like this
availa
Tilghman Lesher wrote:
On Monday 22 December 2003 10:12, Philipp von Klitzing wrote:
Hi!
I'm also curious if anyone else is doing this or if anyone else
is using the Asterisk TDD support.
Excuse my ignorance: What exactly is TDD? Is it US specific?
It's a specification for sen
Walker Haddock wrote:
On Mon, Dec 15, 2003 at 05:06:57PM +0200, Dan wrote:
Hi,
- Original Message -
From: "Walker Haddock" <[EMAIL PROTECTED]>
Dan, you say fax works better on the TDM400 than the ATA186. I'm having
problems with faxing on the TDM400.
I had to drop t
ProvoCityPower wrote:
The question asked here, "why on earth you want to push fax data over
a VoIP link at
all. Fax compression isn't very efficient." may speak volumes about
the future role of VOIP. My plans are to role out a VOIP connection to
thousands of Customers. Many have legacy fax equi
Alastair Maw wrote:
Fax uses FSK modulation to transmit the data. If you compress this in
a lossy way (GSM, MP3, whatever) then the integrity of the data is
affected (more or less seriously depending on the codec used). Fax
machines are generally quite picky, so compressing faxes is unlikely
t
Hi all,
Does anyone know of any work in progress on IAX based telephony for
PDAs? Putting iaxcomm on a Zaurus or iPAQ, for example.
Regards,
Steve
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Dan wrote:
Hi Steve,
- Original Message -
From: "Steve Underwood" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Sunday, December 07, 2003 10:36 AM
Subject: Re: [Asterisk-Users] RxFAX application
Hi Dan,
Dan wrote:
Hi,
I have started the * server in
Hi Dan,
Dan wrote:
Hi,
I have started the * server in console mode (-vc) and this is what I get
now (no file saved and disconnected):
[]
It seems the software FAX modem is sending out its messages regularly,
but never hears anything recognisable come back from the far end FAX
machi
Tristan 'Minty' Colgate wrote:
Hi,
My apologies for those on the channel who may take offense to this, atleast
the ones to whom it was not aimed, but the fact is that after making a simple
enquiry on the IRC channel I was in absolute shock...
I asked one simple question, are there any known iss
John Todd wrote:
Obviously, there are no DS3 TDM cards that are currently compatible
with Zap channels. (or are there?)
Does anyone know of an inexpensive DS3 card that could perhaps be used
with Asterisk if one were to try to port the Zap drivers to such a
card? PCI, of course, would be the
Roy Sigurd Karlsbakk wrote:
Just to inform the community ... i received an
offer last week for 1 week of asterisk training
+/-2USD !! We can't aford this !
Is that USD 20.000,- as in twenty thousand US dollars, or is have
someone played around with the keyboard? If so - who the fu
Brian Capouch wrote:
Michael Devenijn wrote:
We are working with realspeak and it is a wonderfull product (even in
product) it supports up to 20 languages and has aquired a really good
prod. stability !
What kind of money we talking for that product?
I think i
Carl Youngblood wrote:
What is EAGI? I will probably use festival for the time being, but
I thing that I would eventually like to use ScanSoft's RealSpeak SDK
because it is so life-like. Unfortunately our text alerts are fully
customizeable, so we can't pre-record them.
Beware the likeli
Carl Youngblood wrote:
What is EAGI? I will probably use festival for the time being, but I
thing that I would eventually like to use ScanSoft's RealSpeak SDK
because it is so life-like. Unfortunately our text alerts are fully
customizeable, so we can't pre-record them.
Beware the likelike TT
Cristian Vasiliu wrote:
Why PCI 3.3V for E1/T1 card!? I can not use it because I can not find
any motherboard with PCI 3.3 . Any sugestions!?
You have four options:
A 32 bit slot in a Dell 600SC
Almost any 64bit PCI slot (except for a small number of 33MHz only 64
bit slots)
Wait for the soon
Steven Critchfield wrote:
On Tue, 2003-11-25 at 09:24, Jerimiah Cole wrote:
Does anybody have useful tips on creating good quality recordings for
use with prompts in asterisk? I'm interested in hearing input on
hardware (mics, dats, sound cards, etc) and software (recording
software, dsp) a
Isamar Maia wrote:
Why don't you put it on the -dev list. Even if most of us might not be
able to help much, we could watch and learn.
The issue to create several lists was already decided and implemented?
If so, let me know since I didn't get this thread.
I implemented routines to analyse
mattf wrote:
Hello,
Ok, I've tried tinkering with the ring voltage (this is an editable variable
on the Sipure SPA-2000) and I still can't get it to work. I plug a generic
POTS line in and it works, I plug an analog port from our old Comdial PBX in
and it works.
with the sipura adapter I've tri
Hi Dan,
From the log, it looks like something wrong is in the modem software. I
have been adding a V.27ter modem to the FAX software, to do the slow
2400, and 4800 bps FAX modes. I've been busy with other things the last
couple of weeks, but I am finishing this off now. I have also made the
e
Steven Critchfield wrote:
Callerid in the Us is signaled with FSK between the first and second
ring. For more information, you will need to consult the source as it
will be definitive.
So far, I think it has been established that in the UK, the callerid is
sent after a line reversal and before th
Hi Antonio,
This is often a pain with ISDN. What works varies from place to place.
Ahm the wonders of standards :-). Setting the dial plan to international
is probably right. When you do this you may need to drop the 00 prefix,
and start with the country code. Then again, you may not. It varies
Anthony Wood wrote:
On Wed, Nov 19, 2003 at 10:22:55AM +0800, Steve Underwood wrote:
Arnold Ligtvoet wrote:
Since I would like the user names to be auto-generated by the system, I
would guess that this could best be done using festival with a localized
voice. I think there is a Dutch
Arnold Ligtvoet wrote:
Since I would like the user names to be auto-generated by the system, I
would guess that this could best be done using festival with a localized
voice. I think there is a Dutch voice for Mbrola with should integrate into
festival ( note to self : need bigger harddisk :-) )
Andrew Kohlsmith wrote:
Oh and BTW, UTFG before asking a question. You would have probably seen
... UTFG?
May the Google be with you, always :-)
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Hi,
I've kind of ported a DTMF text extry method I wrote some time ago for
Dialogic. It is now a semi-working Asterisk app. I've still got to clean
up some stuff in how Festival is used to read back what is entered, and
then I think it should be OK.
Would anyone here find this useful? It takes
Engineers of all kinds can be a bit lax about documentation. However,
the documentation police are rightly held in a regard usually reserved
for lawyers, realtors, used car salesmen and serial killers.
There isn't a single thing to stop anyone that really loves
documentation actually producing
Stephen R. Besch wrote:
God, I wish I could be so sure of that. I've looked at the circuitry
on some high end channel bank active hybrids. SPICE modeling predicts
a maximum 26 dB attenuation of the returned echo, even with a balanced
line. A simple circuitry change which adds a balance adjust
Stephen R. Besch wrote:
5) Attempt to balance the hybrid at the 2-line to 4 line interface.
Why: 99% of the time, this is where the echo originates
and this is where is should be fixed. Unfortunately, this is not for
the faint of heart, but if your line card has a hybrid balance
Andrew Kohlsmith wrote:
An E1 can be a long way from the box with the right cable. However many
people use the wrong cable. Using a LAN cable for an E1 often gives
errors if the cable is more than just a few metres long. Although the
plugs look the same, the twisted pairs should be grouped differe
Ariel Batista wrote:
Can anyone think of any others?
Here is the short list I have!
1) Lack of graphical interface.
2) Un-freindly user interface (Command prompt only)
3) Network and Telephony person needed at site.
4) No standard SIP Phone nor IAX phone available.
and the biggest one I fe
WipeOut wrote:
Gavin Hamill wrote:
It would seem an odd question, but I'm trying to put together a little
presentation on 'Why Asterisk?' and need to list Pros and Cons I've
got plenty of Pros (including the availability of commercial support),
but the only Con I can think of is 'Relatively f
An E1 can be a long way from the box with the right cable. However many
people use the wrong cable. Using a LAN cable for an E1 often gives
errors if the cable is more than just a few metres long. Although the
plugs look the same, the twisted pairs should be grouped differently in
an E1 cable,
Andrew Gillham wrote:
Steve Underwood wrote:
Hi Thomas,
Unless you have a *very* specific need to use G.723.1 for
compatibility with someone else, forget it. It is pretty much an
obsolete product. Licencing is also a pain, as there is not patent
pool for it. G.729 is expensive to licence
Hi Ernest,
I tried IPP, but couldn't get much performance out of it. When I tried
diassembling one or two routines to see what they looked like, there
seemed at be a llo of overhead in the routines that
destroyed all the benefits.
Regards,
Steve
Ernest W. Lessenger wrote:
Hey all,
Hi Thomas,
Unless you have a *very* specific need to use G.723.1 for compatibility
with someone else, forget it. It is pretty much an obsolete product.
Licencing is also a pain, as there is not patent pool for it. G.729 is
expensive to licence, but at least it is relatively strightforward. If
r at
least, expected to be much further away than an extension phone, I was
wondering if there's
a difference in the electrical requirment.
thanks again,
patrick
Steve Underwood wrote:
Hi Patrick,
You are in the UK, right (at least DDI strongly suggests that)? This
is the commonest signalling
Hi Scott,
I use a Tyan 2665 (7505 based) M/B with a TE410P. That works well. This
is a development workstation, so its probably not the kind of board you
want for deployment.
Regards,
Steve
Scott Stingel wrote:
Hi-
I'm looking for an appropriate system board to power a system with two (2)
Dig
Hi Patrick,
You are in the UK, right (at least DDI strongly suggests that)? This is
the commonest signalling for a DDI line on an analogue pair. The line is
behaving just like the main exchange is a telephone. It picks up the
line, by applying a 600ohm loop, and dials (with pulses per second or
Ray Burkholder wrote:
Might want to write a new
"energy detector" algorithm in dsp.c though based on a wideband/low Q
resonator approach (move the pole way in towards the origin)
as opposed to
narrow band goertzels (pole on the unit circle). More robust
for this type
of work.
Where d
I am taking note of people's messages about soft fax, even if I might
appear to be ignoring them. I am getting V.27ter finished off right now,
to flesh out the facilities in the software. V.27ter is used for 4800bps
and 2400bps faxes - not critically important, but useless for lousy
lines. That
Hi Chris,
That is exactly the Dialogic implementation I was referring to that was
utterly useless. It works OK when people are demoing, as they always
follow a certain pattern. In real like it I've always found it a recipe
for screaming angry users. Depnding on your use it can get over 90% of
Did you mistype or something. That link is about power profiling the
consumption of DSPs :-)
Regards,
Steve
Asterisk online forums wrote:
some information can be found here about algorithm and descriptions of
method being used.
http://citeseer.nj.nec.com/393112.html
Regards,
Alexander
**
Lists wrote:
On Tue, 28 Oct 2003, Brian Schrock wrote:
Everyone,
Just thought I would drop a line telling everyone here I have the software
RxFAX/TxFAX up and running without any real problems. I did have to.
RH 9.0
1) Install an audio devel rpm
1) install libtiff from source, and copy
Alastair Maw wrote:
On 27/10/03 21:57, DUSTIN WILDES wrote:
Does anyone have any recommendations on implementing Answering
Machine detection for call generation programs?
There's obviously no nice way of doing this.
If you're doing telemarketing, and you're playing pre-recorded audio,
which of
links
in ITP, how load a SPC formats, and how can I add an TGCP route in BTS...
Sure! Why not?
Regards,
Gus
- Original Message -
From: "Steve Underwood" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Sunday, October 26, 2003 2:14 AM
Subject: Re: [Asterisk-Users] SS
Interesting. Someone thinks that a strategic use for * should be off
this list. Someone thought my FAX modem for * should be off this list.
However, nobody seems to think a 1000 messages about Grandstream phones
should be off this list.
Personally I would welcome seeing more of what people are
Steven Critchfield wrote:
[...]
Now I just wish it was robust enough to work over my IAX connection
without problem. I'm sure it is the GSM compression that is biting it.
I'll run some more tests and then maybe I'll move it to a machine
located in the same rack as my phone gateway machine. At tha
Hi Steven,
I think I have the tagging right for the aspect ratio. A lot of display
software gets it wrong, including some well regarded things like the
GIMP. KFax displayed my fine and standard test FAXes properly.
Steven Critchfield wrote:
Figured the group would like to hear this. I just fax
Hi Chris,
What on earth are you refering to?
Regards,
Steve
Chris Albertson wrote:
This whole argument is moot because there IS a free g.729
implementation. Actually it is a zero cost license to the
source code. Exactly what was asked for.
--- Steve Underwood <[EMAIL PROTECTED]>
)
Comments are welcome.
Thank you.
Alex Zarubin
-Original Message-
From: Dave Cotton [mailto:[EMAIL PROTECTED]
Sent: Tuesday, October 21, 2003 11:45 AM
To: Asterisk List
Subject: Re: [Asterisk-Users] A software FAX modem
On Tue, 2003-10-21 at 16:22, Steve Underwood wrote:
> I did say this wa
Witold Krecicki wrote:
1st. - I'm from Poland, we don't have (yet, and hopefully forever) software
patents.
Is there any free g.729.1 implementation for asterisk? I want to use it for my
private use (dialing into inet->PSTN gateway), and I don't want (now) to buy
codec, as I don't know if I wi
). I also fixed something that could cause crashes if you tried to send
a non-existant TIFF file.
I'm sorry if I disappointed the early adopters, but it *will* get better.
Regards,
Steve
Florian Overkamp wrote:
Hi,
Citeren Steve Underwood <[EMAIL PROTECTED]>:
If it doesn't w
Hi Florian,
Florian Overkamp wrote:
Hi,
Citeren Steve Underwood <[EMAIL PROTECTED]>:
If it doesn't work for you, don't be too surprised. Feed back anything
you find, and lets try to make things better. I suspect, from experience
and things I have read on the web, that a lot
Hi all,
I would like to announce the availability of an initial test version of
a totally software FAX facility, suitable for use with Asterisk. This is
a first public test release, so don't expect a solid polished product
just yet. People have shown interest in what I am doing, and here is the
Klaus-Peter Junghanns wrote:
Hi Martin,
libpri misses all the fun stuff :-(
hold, retrieve, suspend, ect, cd, conf, 3pty ..
but i am going to change that :-)
regards
kapejod
It misses all the timers, too. :-)
Regards,
Steve
___
Asterisk-Users m
TC wrote:
Mythical Asterisk Creatures, oft-discussed, rarely seen:
1) An "advanced" graphical user interface
2) An IAX2 hardware device
3) A Radius CDR report module
4) A live-method, robust SQL-based dialplan
5) LDAP/SQL/Radius authentication for SIP phones
6) Robust R2 signalling suppor
Hi Brad,
If you want to detect that a sound is voice, rather than something else,
it isn't easy. There is information around on the Internet about
methods, but I have never tried them and don't know how well they work.
Unless you have some understanding of DSP I wouldn't bother trying. On
the
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