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
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
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 cannot
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
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 country where the key
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
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
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taken a step up - kind of sad it should improve now it is obsolete :-)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
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.
ATT said that about teletype service, you know 50 -
300 baud service,
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:
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?
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
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
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
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
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
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
irc://freenode/%23asterisk
What's up?
Regards,
Steve
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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 channel on IRC has become an exclusive
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
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
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
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
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
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
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]
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
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
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
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
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 box unless you
[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
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,
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
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
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
Paulo Mannheimer wrote:
Hi,
I'm testing an E1 with EM signaling. Some of the problems I'm running
into are the following:
1) if I try to configure any channel above channel 15, I start
getting a multiframe alignment error on my telco test equipment. So I
have my zaptel file only configured
Hi Martin,
Strange. After I fixed the bugs that screwed up the top half of the
timeslots in CAS mode, I had no trouble with my E400P framing. I used it
with and without CRC4, and had no trouble of that kind. I had some weird
stuff with an E100P, but I think that was something to do with the
Do people actually do the *ANI*DNIS* thing on E1s? I've never seen that.
E1s are a real pain for anything but PRI or SS7. There is so little
standardisation. A place I used to work has a substantial team turning
out new signalling protocol state machines for each customer of its E1
muxes.
, and compile them directly down to ROM tables). Either way you
build a state machine.
Regards,
Steve
Martin Pycko wrote:
Maybe if they'd write the PRI stack in C instead of making a state machine
they woun'd need to make adjustments so often.
regards
Martin
On Wed, 3 Sep 2003, Steve Underwood wrote
Paul Lambert wrote:
Not yet. implies that it is coming.
Look at the latency it causes, and you will see its not that useful.
I know it would help on Internet
connections such as fixed wireless and cable modem where packet rate is
an issue. 20ms translates to 50 packets/sec.
30ms per block
That is not just true of IAX. There appears to be substantial amount of
RTP traffic, which trunks a variable bundle of calls between the same
two points, used by IDD services. The traffic has to be going between
the same two points to make that work, though, whichever protocol you
use as the
Azher Amin wrote:
Hi,
Can anyone suggest a good motherboard for the T/E410P cards ? Coz it
doesn't get inserted in the the regular P4 motherboards due to PCI
slot (32 bit) Any suggestions.
Regards
Azher
Do you
Alex Zarubin wrote:
I am positive, 4 bits per sample, 6000 Hz.
This is a default play/record setting for the older Dialogic R4 API
and we need
to play zillions (sic!) of files (messages) recorded this way.
Conversion issues:
- expensive
C versions of the OKI/Dialogic ADPCM codec are
If shorting two FXS lines together damages them they are badly designed.
Good BORSCHT (battery, over-voltage protection, ringing, signaling,
hybrid, and test) design should mean they can tolerate this kind of
thing. They have to very often in the poorly controlled PSTN rats nest.
Regards,
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 support
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
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 of fax machines
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 work for you, don't be too surprised. Feed back
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
-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 was a first test release :-) I can't be held
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] wrote
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
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
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
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] SS7 signaling/Softswitch
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
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
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
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
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.
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
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
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)
, 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 for a DDI line
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
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
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
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,
Martin List-Petersen wrote:
On Mon, 2004-11-15 at 07:46, Talbot Neil wrote:
Hi,
I was wondering if there is any PRI hardware that is Austel certified
and works
well with Asterisk under FreeBSD???
If anyone has any information please let me know as I seem to be
having problems
finding any
Hi Mike,
SS7 connects to the * box, with no other boxes next be involved. Of
course, that box could be configured as a gateway, and act just like
many of the SS7 converter boxes.
Regards,
Steve
Mike Machado wrote:
Have you implemented some kind of device control protocol to a gateway,
or do the
Hi Angel,
It is working pretty well. I think it will be available about the end of
the year. I will not be free. It will be supplied with a commercially
licenced Asterisk.
Regards,
Steve
Angel Diaz wrote:
Hi all,
Does somebody know what's new with SS7 and * ?
I'm very interested. Is it ready
Hi Leonardo,
This is not a libtool issue. It looks like you must have an ancient C
compiler, that doesn't understand C99 constructs.
Steve
Leonardo Gomes Figueira wrote:
Hi,
Trying to update to spandsp-0.0.2-pre6 I got a compile error:
Making all in src
make[1]: Entering directory
Leonardo Gomes Figueira wrote:
Steve,
Steve Underwood wrote:
Hi Leonardo,
This is not a libtool issue. It looks like you must have an ancient C
compiler, that doesn't understand C99 constructs.
gcc 2.95.3
Any workaround or I really need to upgrade gcc ?
Leonardo
That's really old. Right now
Matthew Crocker wrote:
I just avoid people who think it's ok to create proprietary extensions
to free software. People like that should be ashamed of themselves,
as it's just an insult to the people who have freely contributed to
the project.
I fully agree.
How hard would it be to integrate
Dinesh Nair wrote:
On 19/11/2004 21:30 Steve Underwood said the following:
I can't imagine anyone successfully integrating openss7 into
anything. I believe it works OK on its own, and is in use as a
gateway. It wasn't
as a gateway between what ? if it's SS7 on one side, what's on the
other
Dinesh Nair wrote:
On 19/11/2004 22:44 Steve Underwood said the following:
as a gateway between what ? if it's SS7 on one side, what's on the
other ? SIGTRAN (SS7 over IP) on top of SCTP ?
Yep, that kind of gateway. He has his own SCTP, and doesn't use the
native Linux 2.6 one.
in which case
Kevin P. Fleming wrote:
Eric Hall wrote:
Telco gives me a PRI It connects to my Asterisk via Wildcard T100P using
a 2nd Wildcard T100P I would like to connect with a X-Cable my HylaFax
server.
My question is will a Wildcard T100P work in a Hylafax server?
That depends on your definition of work.
John Hill wrote:
Trying to send a fax using a call file and txfax.
Phone dials the remote fax answers but * gives me:
Call failed to go through, reason 3
And hangs up.
Any help.
Thanks
--John
Did you add |caller to the command?
Steve
___
kaws elchamal wrote:
I patch asterisk in this way:
patch - p1 channels_makefile.patch is it the good way to patch
it?
after the patch, I complie asterisk without problem I configure the
unicall.conf as needs. compiling asterisk patched dont generate any
error but the file
HO SIN wrote:
Dear Kaws, and Dear Steve,
I think I am facing the same kind of trouble. This kaws's case is
simple; one line inserted by patch designates wrong header file name
of unical.h, therefore the objects are not compiled. The reality
is unicall.h, and
OK, fixed that one.
after
Peter Svensson wrote:
On Fri, 26 Nov 2004, Andrew Kohlsmith wrote:
There can be only one clock and you must engineer your system such that
everything is synchronized properly. For simple systems like what we are
describing it's not difficult but when you have multiple spans coming from
Corvin wrote:
Adjust your expectations to telephone quality. Everything is based
around 8khz samples and at best around 14 bit quality. The GSM files
should be around cell phone quality due to the codec. You are welcome to
rerecord any sound prompts you wish and use them at a slightly higher
Nicolás Gudiño wrote:
Hello,
I'm unable to compile testcpuid.c with the __x86_64__ architecture
(Athlon 64 processor). The messages are:
/tmp/ccONleRV.s: Assembly messages:
/tmp/ccONleRV.s: Error: suffix or operands invalid for 'pushf'
'pop'
'push'
'popf'
Is it safe to ignore this module?
Darren Nickerson wrote:
James H. Thompson [EMAIL PROTECTED] wrote:
T.38 is often put forward as the solution for reliable FAX over VOIP.
With good reason ;-)
Just wondering for anyone using T.38 (with any equipment), how well does
it work as compared to a FAX PSTN call?
Our experience is that
Guilherme Góes wrote:
Does anyone have the fixed point implementation of the iLBC codec ? I
am an undergraduate student that's trying to implement a faster way
for this CODEC in a PC architeture using MMX/SSE (fixed point).
On a PC it is actually quite hard to make a codec like this go faster
Patrick wrote:
Dear List,
I'm running an Asterisk 1.0 server with 4 HFC cards and bri-stuff behind an
Anlagenanschluß with 8 B-channels in Germany. It worked fine with Deutsche
Telekom, but since we switched to Arcor nothing works at all.
After some debugging, I called Arcor helpdesk who told me
Peter Svensson wrote:
On Wed, 1 Dec 2004, Steve Underwood wrote:
Patrick wrote:
I'm running an Asterisk 1.0 server with 4 HFC cards and bri-stuff behind an
Anlagenanschluß with 8 B-channels in Germany. It worked fine with Deutsche
Telekom, but since we switched to Arcor nothing works
WipeOut wrote:
Michael Vogel wrote:
Hi!
Some - few - providers are using IAX2 as a protocol. Most are using
SIP. I know that there are advantages of IAX2 regarding multiple
connections. But beside this I'm asking myself (and you all) why I
should prefer IAX2 when my SIP connection is working.
Hi Scott,
Watch out for spandsp-0.0.2pre7 :-)
Seriously, I found the same issue while testing for x86-64 machines this
week. spandsp is now working on a x86-64, but I haven't tidied up and
released an updated version yet. That should happen this weekend. I hope
this updated version will build
Roy Sigurd Karlsbakk wrote:
hi all
according to what I've found out this far, the G.729 patent seems not
valid in a broad range of countries.
Don't bet on it.
does anyone know where I can find the algorithm?
Its called G.729 because it is defined in the G.729 spec. :-\
Steve
Albania, I think :-)
Steve
Robert Rozman wrote:
Hi,
do you have info in what countries g.729 is not valid... ?
Regards,
Robert.
- Original Message -
From: Roy Sigurd Karlsbakk [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]; Asterisk
Guilherme Góes wrote:
Does anyone have the C reference code of the ITU G.711 Appendix II ?
I don't think there is specific reference code for this. However, if you
read the appendix you will find it is just using a section of the G.729
algorithm. You can lift suitable code out of the
Kevin Walsh wrote:
Robert Rozman [EMAIL PROTECTED] lazily top-posted:
do you have info in what countries g.729 is not valid... ?
You could start with the whole of Europe and can also add the UK.
I'm sure there are lots of other countries who don't feel the need to
acknowledge US-based
Eric Wieling aka ManxPower wrote:
Steve Underwood wrote:
Albania, I think :-)
Cite your source.
I might be wrong. I'm working from second hand knowledge. Someone told
be they never introduce copyright legislation and their patent
legislation is almost non-existant. I think you would
Hi Matthew,
See http://www.opencall.org/faq/x47.html
Regards,
Steve
Matthew Boehm wrote:
Here is the setup:
POTS - PRI - Asterisk - ATA (Fax)
The ATA is set to only 711. Asterisk's sip.conf sets this device to only
711. Yet, faxing works less than 50% of the time.
I cannot possibly be the only
Andrew Kohlsmith wrote:
On December 7, 2004 04:30 pm, Matthew Boehm wrote:
Yes, using RedHat 9.
Are you using RH's stock kernel or a plain-vanilla kernel? I have heard
nothing but bad things with Asterisk and RH's custom kernels. If you can,
try a stock 2.6.9.
It is just the
If you really added /var/local/lib, that would eb your trouble. It
should be /usr/local/lib.
Steve
Ariel Batista wrote:
I have compiled Spandsp without any problems. I got no errors I have
also done the patch without getting any error. I have tried pre4 and
pre6 version with same problem. I
Fabrício Zimmerer Murta wrote:
Oh, friend... I have realised just yesterday that's impossible to use
regular modems (say hayes/v90 33.6 or 56k) to plug asterisk to the world. I
can't figure out why. But they simply don't support it.
If you want to use your isdn modem to plug * to the world, it's
Michael Vogel wrote:
Hi!
The encoding, decoding and recoding cost cpu time, that's sure. But
does this time differs much depending on the used codec?
Is - for example - a G729 faster than a GSM codec?
Bye!
Michael
They vary a lot. G.729 is pretty slow. iLBC and speex in the same
ballpark.
Rafael Pazetto wrote:
Hello,
Any chance of development team make more drings?I googled and found out
that only 3 drings can be configured.I have at least 6 different rings
in my lines.In my setup is a good feature since I can know whos coming
from the world and whos coming from inside the
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