Roland Welker wrote:
Hello,
Does anyone now, if there are any legal requirements for setups of
Digital (i.e. Linux/Asterisk) based PBXs in the UK? I am especially
interested, if a system does need to hang on a UPS?
Unless things have changed, you need either:
- 7 hours of UPS support or
Leandro Morgado wrote:
Steve Underwood wrote:
Robert Rozman wrote:
Hi,
I'm getting unreliable dtmf recognition (it works fine for 4-5
digits, errors (duplicates) on more), when transferred inband from
gsm gateway to NT port of quadbri under bristuffed Asterisk.
Since Ast
Peter Svensson wrote:
On Tue, 21 Jun 2005, Leandro Morgado wrote:
Steve Underwood wrote:
Robert Rozman wrote:
I'm getting unreliable dtmf recognition (it works fine for 4-5
digits, errors (duplicates) on more), when transferred inband from
gsm gateway to NT port of qu
Robert Rozman wrote:
Hi,
I'm getting unreliable dtmf recognition (it works fine for 4-5 digits,
errors (duplicates) on more), when transferred inband from gsm gateway
to NT port of quadbri under bristuffed Asterisk.
Since Asterisk is claimed to have good dtmf recognizer, I suspect
ther
j_amorim wrote:
Hi,
MFC R2 - UniCall implementation.
The * is configured to send a 1101 Idle signal:
zaptel.conf
span=3,1,0,cas,hdb3
#
cas=9-23:1101
cas=25-39:1101
You seem to have configured 1101 as the blocking signal.
But is sending 1001 Idle signal
1001 is the usual id
Lee Howard wrote:
Marco Parmeggiani wrote:
Can someone explain me what's going on and why the receiver of this
fax guives up saying communication error?
Start tx page 1
>>> EOP: 2f
<<< RTN: 4c
>>> DCN: fb
Disconnecting
The receiver says communication error because txfax's response to the
They keep breaking the FAX support in libtiff. 3.6.1 is broken, although
a lot of distributions contain a patched version which works - usually
because a spandsp user got the patch pushed into the distribution.
HylaFAX users seemed to just give up trying to follow the buggy path of
libtiff, and
j_amorim wrote:
Hello guys,
I am having problems to installing libsupertone library.
The ./configure --prefix=/usr does not create the Makefile.
Any tip?
It has worked for everyone else. Lots of people have successfully
employed this library.
Steve
___
Script Head wrote:
I have minor echo on an IAX2 channel when using Firefly and a head
set. I have tried various headsets and settings but still a little bit
of the echo remains and I'd love to get rid of it. After some research
I stubled on zaptel/mec2.h but it seem that it works only on the ZAP
Dave Cotton wrote:
With the current CVS-HEAD line 88 of app_rxfax.c causes an error.
#if (ASTERISK_VERSION_NUM <= 010300)
chan->callerid,
app_rxfax.c:88: error: 'struct ast_channel' has no member named
'callerid'
Commenting out the if else combination of course gives a cl
Actually, Intel have no right to ake any claims about what you can do
with that code. They are neither the copyright holder, nor the patent
holder. If you look at the code it is just the ITU reference code
modified to take advantage of Intel's IPP library.
Steve
stevanus wrote:
Hi,
As far
contain the fix.
Regards,
Steve
Andres Maduro wrote:
Hi,
I have recently found a bug when using Steve Underwood chan_unicall
with Asterisk 1.0.x (including 1.0.8RC)
When you place a call from a SIP phone with dtmfmode=rfc2833 or
dtmfmode=inband through MFCR2 via chan_unicall all goes
Actually, they are compressed, but they are free to use :-)
Steve
Sahil Gupta wrote:
Hi,
Both of those are fully uncompressed codecs and free to use.
Regards,
Sahil Gupta
VoiceValley
On Fri, 10 Jun 2005, Edgardo Bermejo wrote:
Hi,
Its possible to make a pass-trhu conection with alaw o
Gavin Hamill wrote:
On Sunday 05 June 2005 16:31, Chris Mason (Lists) wrote:
I have purchased 50 licenses at $10 each from Digium,
Cool, so you have satisfied yourself that you are licensed to use the G.729
codec and not get your ass sued by the IP holders. Now you can simply use th
Andrew Kohlsmith wrote:
On Sunday 05 June 2005 09:27, trixter http://www.0xdecafbad.com wrote:
That was a policy we did not adopt, something about using the word
'unlimited' and then not wanting to fill it with a ton of qualifiers
like 'its unlimited unless you actually use then then we will
Tom Fanning wrote:
Agreed, those are the figures we were able to get
from Digium... I'm still waiting for a confirmation,
but I'm being safe with a $4k estimate..
What's so special about Digium cards that makes them this expensive? $4000
for a PCB is extortion IMO!
I'm sure
Jorge Alayon wrote:
Hello all,
After compiling successfully Asterisk and AMPortal, I cannot make the fax
module work.
Asterisk does not start (unless I remove the modules or mark them as Noload
in modules.conf) with the following error:
Jun 3 20:55:25 VERBOSE[3328]: [app_rxfax.so]Jun 3 20:
Script Head wrote:
I was pondering of the best way to implement voice-coloring within
Asterisk, e.g. pass a channel thru a multiband equalizer and modify it
enough where it could be distinguished from other voices in a
conference call. This could make conference calls much less confusing.
Perha
Hi Marcin,
You need to turn verbose logging on to see what really happens. Check
the command line parameters for rxfax. Lots and lots of people have
problems faxing with TDM cards. The problem is not in spandsp. It seems
to be in the river for the card.
Regards,
Steve
Marcin Kuczera wrote:
Tony Mountifield wrote:
In article <[EMAIL PROTECTED]>,
Andrew Kohlsmith <[EMAIL PROTECTED]> wrote:
You can convert between ulaw and alaw 1 times and it'll sound exactly the
same as the original 8000Hz 16-bit sample because... well... it is.
Just to pick a nit, the above can't be
[EMAIL PROTECTED] wrote:
On Sun, 29 May 2005, Steve Underwood wrote:
It isn't a one way scale. Music on hold over G.729 is often
unrecognisable. Over GSM 06.10 it is usually just poor. Is that a big
issue for you, or totally irrelevant? GSM 06.10 and G.729 at 8kbps offer
fairly si
Marie wrote:
Anyone have the time and webspace to post a quick recording of a
sample conversation in both codecs? If you want to get even more
tricky, perhaps samples of music on hold in both as well? Or noisy
environments?
This kind of quickie test is worthless. In doing serious codec
evalu
It isn't made Soyo. Its an ODM job from one of the makers in Shenzhen. I
forget which, off hand. Hunt for that, and it should tell you which of
the standard PA168 firmwares is right for it.
Regards,
Steve
Isamar Maia wrote:
Yes. it's a PA1688. IMHO, it works well for Home users but don't ev
Hi,
It looks like you are using Asterisk 1.1.x. Until yesterday
chan_unicall.c only worked with 1.0.x. In the 0.0.3pre2 directory you
will now find versions of chan_unicall.c for Asterisk 1.0.x and 1.1.x.
The version for 1.0.x is well tested. The version for 1.1.x is new, and
only slightly te
Tim Pushor wrote:
Its obvious that Steve never looses, even when he's wrong, so arguing
about it to him won't get anywhere.
I have been known to loose arguments. Not too many, but some. :-)
As for g729, I was pleasantly surprised by the quality.
I fully agree.
I may be old fashioned, b
Great log. You carefully stripped out anything that might possibly be
useful for diagnosise. However, the answer will be the same in any case
- See http://www.soft-switch.org/foip.html
Regards,
Steve
Zen Kato wrote:
I chaged to use from 'rxfax' to 'txfax' and I succeeded to receive the
file
Nguyen Trung Tin wrote:
Hello All
I need to use Asterisk with an E1 sangoma card with CAS R2 signalling
for Vietnam
what is difference between libr2 of CVS and libmfc2 of soft-switch.org ?
libr2 is a piece of useless junk, which I have asked Digium to remove.
The software at soft-switch.
Michael D Schelin wrote:
Steve, you should really test the Codec and have G729 running as a
pure IP to IP call you can not hear the difference on good networks!
Well, it does to anyone without hearing damage. It sounds very obviously
different.
Please do not get me wrong that G711u sounds
Michael D Schelin wrote:
I have used G729 and it sounds almost as good as G711U. The problem is
the way Asterisk uses it. It does not sound robotic and it's not
suppose to sound that way. Most Carriers want the calls to be in
g711u so thats why I use G711u otherwise I want to save money on
b
chawki hammoud wrote:
I installed G729 from Diguim and I was expecting the
sound quality on my i686 machine to be better than
gsm. Compared to gsm, G729 sounds closer and a little
robotic. Is this what is supposed to be or am I
missing something?
I am interested in G729 because the internet in
- Original Message - From: "Steve Underwood" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, May 25, 2005 5:14 AM
Subject: Re: [Asterisk-Users] G729 codec
Ivan Meic (Vox Mundi) wrote:
Actually G.729A is a reduced comp
Andrew Kohlsmith wrote:
On May 25, 2005 10:02 am, Steve Underwood wrote:
So let me get his clear. If I don't document things I am in the wrong,
whereas if I do document them I am in the wrong. Is that it?
No no; I have to admit that I took the foip page as a background on faxin
Rich Adamson wrote:
Steve, what would help a bunch of people trying to implement your
spandsp is some kind of help document that at least attempts to
describe some of the debug statements shown below. When the average
person reads "hdlc underflow" or "T4 timeout in state 9", we don't
have a clue
Rich Adamson wrote:
Steve, what would help a bunch of people trying to implement your
spandsp is some kind of help document that at least attempts to
describe some of the debug statements shown below. When the average
person reads "hdlc underflow" or "T4 timeout in state 9", we don't
have a clue
Andrew Kohlsmith wrote:
On May 25, 2005 09:46 am, Rich Adamson wrote:
Steve, what would help a bunch of people trying to implement your
spandsp is some kind of help document that at least attempts to
describe some of the debug statements shown below. When the average
person reads "hdlc under
Hi Zen,
See http://www.soft-switch.org/foip.html
Steve
Zen Kato wrote:
After I did rxfax(...|debug), I got the followings;
..(snip)...
DIS: 80 00 ce f4 80 80 81 80 80 80 18
HDLC underflow in state 9
Changed from phase 4 to 3
T4 timeout in state 9
Changed from phase 3 to
Ivan Meic (Vox Mundi) wrote:
Actually G.729A is a reduced complexity version, and G.729B is a version
with silence suppression. The data rate while sending voice is exactly
the same, although the quality of G.729B should be a little higher.
However the average rate for B can be lower if the si
Actually G.729A is a reduced complexity version, and G.729B is a version
with silence suppression. The data rate while sending voice is exactly
the same, although the quality of G.729B should be a little higher.
However the average rate for B can be lower if the silence suppression
is used. Rig
Rich Adamson wrote:
Anyone have a practical experience/knowledge relative to why a 5ESS
central office switch would require a "w" in the Dial statement to
handle analog pstn-fxo calls?
I fully understand what "w" is doing, just trying to better understand
why a 5ESS doesn't accept dtmf a lit
Rich Adamson wrote:
Anyone have a practical experience/knowledge relative to why a 5ESS
central office switch would require a "w" in the Dial statement to
handle analog pstn-fxo calls?
I fully understand what "w" is doing, just trying to better understand
why a 5ESS doesn't accept dtmf a lit
isk. How to acctually implament this though, I have no idea.
On 5/23/05, Steve Underwood <[EMAIL PROTECTED]> wrote:
Matthew wrote:
Is it possible to use openh323's t38modem with asterisk and spanddsp?
or would hylafax have to be thrown into the mix? If it is possilble
how woul
Matthew wrote:
Is it possible to use openh323's t38modem with asterisk and spanddsp?
or would hylafax have to be thrown into the mix? If it is possilble
how would I go about getting astersik to see it?
I can't imagine any useful way to combine t38modem with Asterisk. Have
you actually looked
Michael Welter wrote:
Michael Stearne wrote:
Hi,
What would you say that the best compression format is for voice
recordings on Asterisk? The tradeoff being the file's size. I like
GSM because of the small files size but the quality isn't great. What
are people finding as a good setting for GSM?
Jean-Yves Avenard wrote:
Hi Peter
On 18/05/2005, at 10:05 PM, Steve Underwood wrote:
It is only there because the sending machine put it there in the
image. Spandsp is not different from how any FAX machine I have ever
used behaves. As well as sending the 20 digit number as text, the
sending
Jean-Yves Avenard wrote:
Hello
On 18/05/2005, at 4:09 PM, Peter Svensson wrote:
I think he is refering to the remote fax id to be presented, not the
header. I.e. the 20 digit user selectable number on the remote fax. The
one often seen on the lcd of the receiving fax and so on.
Yes that's exac
Jean-Yves Avenard wrote:
Hello Steve.
On 17/05/2005, at 10:54 PM, Steve Underwood wrote:
When a fax is received the header you see is part of the image. As
such, it ends up in the TIFF file as part of the image. It is not
available as text anywhere. The only way to make it available as text
Steve Kennedy wrote:
On Tue, May 17, 2005 at 10:45:52PM +0800, Ronald Wiplinger wrote:
Skype is very succesfsfull and get more and more users, ... we can
ignore them, accept them or do something,...
My suggestion is that we try to do something, ...
If we would peer to each other, than we get so
Lee Howard wrote:
On Tue, 17 May 2005, Jean-Yves Avenard wrote:
Side questions about spandsp... Is it possible to print the fax
header like what most faxes do (that is: who is sending the fax, how
many pages are included etc...) I'm not talking about printing
callerid, often I receive fax
"identifier" - a 20 character string, which the standard says should be
digits, and which is usually set to the telephone number of the FAX machine.
Regards,
Steve
Jean-Yves Avenard wrote:
On 16/05/2005, at 11:48 PM, Steve Underwood wrote:
How can that work? You can measure the error, but y
It doesn't help at all, since you are talking rubbish. Try to keep track
of the subject matter. We are discussing modems, where not slipping is
vital.
Regards,
Steve
Rich Adamson wrote:
It doesn't make any difference. The pcm data that arrives from the telco
is buffered in the zaptel and/or aste
Peter Svensson wrote:
On Mon, 16 May 2005, Steve Underwood wrote:
It is possible, though complicated, to synchronize the 2Mbit clocks on two
unrelated cards by measuring the accumulated phase shift (difference in
interrupt rate) over time and compensating, thus implementing a PLL in
software
Dean Collins wrote:
Can we get this looser bumped, this has been happening for the last 2
weeks now.
I hate this kind of thing as much as anyone, but isn't bumping him off a
bit extreme? :-)
Regards,
Steve
-Original Message-
From: [EMAIL PROTECTED] [mailto:MAILER-
[EMAIL PROTECTED]
Sen
Peter Svensson wrote:
On Mon, 16 May 2005, Michael Welter wrote:
Where is the clock source that the T1/E1 board, with "0" for timing,
uses to generate the tx data stream? Is there a PLL on each board? Or
is some central source used?
For example, I have one system with two separate T100P car
Rich Adamson wrote:
I need to connect up to sixteen phones per building, I can use a cheap hub,
but POE would be useful. Is there a cheap POE hub available? Everything I
have seen has been expensive.
Hope you really meant a cheap switch... you don't want to use hubs
of any sort in the asterisk
Dean Collins wrote:
Yep, POE has turned out to be a real fizzer.
Whilst a great idea for Access Points (particularly ceiling mounted AP's
They are *far* more useful for simplifying phone wiring.
so you don't need to run power points) but apart from that the whole
concept has just died.
Not re
Rich Adamson wrote:
What is in the bug tracker helps make things clearer to people who know
what they are doing. What we need is something that makes things clear
to laymen. Saying internally and externally clocked doesn't cut it. It
needs to be made clear to laymen that clocking internally is r
Michael Welter wrote:
What is in the bug tracker helps make things clearer to people who
know what they are doing. What we need is something that makes things
clear to laymen. Saying internally and externally clocked doesn't cut
it. It needs to be made clear to laymen that clocking internally i
Steve Underwood wrote:
Andrew Kohlsmith wrote:
On May 15, 2005 03:46 am, Steve Underwood wrote:
Maybe someone should submit a patch which changes the documentation to
something the average person will interpret correctly. It would have
saved me at least a hundred hours of support hassles in the
Lee Howard wrote:
The next HylaFAX release (current CVS HEAD already has it) will have
color fax receiving support in Class 1/1.0... provided that the
prerequisites are met.
As HylaFAX depends on libtiff for it's image file handling color fax
support with HylaFAX requires that libtiff also supp
Andrew Kohlsmith wrote:
On May 15, 2005 03:46 am, Steve Underwood wrote:
Maybe someone should submit a patch which changes the documentation to
something the average person will interpret correctly. It would have
saved me at least a hundred hours of support hassles in the last year
Jean-Yves Avenard wrote:
Hi
On 15/05/2005, at 5:51 PM, Bryce Chidester wrote:
I have the same trouble with wct1xxp. I'd just chalked it up to a PCI
bug or other low-level hardware problem with the Digium card. I've
simply learned not to, though it would be nice to not have to learn
work-arounds.
Jean-Yves Avenard wrote:
Hello
On 15/05/2005, at 4:40 PM, Steve Underwood wrote:
span=1,1,0,ccs,hdb3,crc4
The second parameter now says "treat this E1 as the first priority as
the clock source". Your box should lock itself to the PSTN's clock.
If that makes no sense, the bottom l
Jean-Yves Avenard wrote:
Hello Steve
On 15/05/2005, at 4:40 PM, Steve Underwood wrote:
The second parameter now says "treat this E1 as the first priority as
the clock source". Your box should lock itself to the PSTN's clock.
If that makes no sense, the bottom line is "this
Jean-Yves Avenard wrote:
Hello
On 15/05/2005, at 1:55 PM, Steve Underwood wrote:
The right thing to do is to sync to the PSTN. The E1s connected to
the PSTN should be listed as the lowest numbered clock sources,
starting from 1. Places you never want to sync to should be set to zero.
Your mail
Jean-Yves Avenard wrote:
Hello
On 15/05/2005, at 12:00 AM, Colin Anderson wrote:
wierd. Im running fc2 2.6.8 smp no problems. Could be timing slips on
your
PRI, happened to me until I looked hard at the PRI
So you turned off Hyperthreading on your linux box or not??
What could I do to check if m
Michael Welter wrote:
Michael D Schelin wrote:
The delay in the air is minor. Radio travels very fast through the
air. Almost at the speed of light. It's the electronics that are
causing the delays. The less electronics touching your signal the
better. The up and down is very fast. But then
Michael D Schelin wrote:
this is beta code! I'm beta testing The t38. Don't use this unless
your testing. It is not backwards compatible.
Please tell me your findings. I am *very* interested in how well T.38
implementations behave. If it total crap, that probably still puts it in
the top 10% :-
Nguyen Trung Tin wrote:
Hello All !
i'm purchased sangoma card A-101. i connect to E1 with MF/R2
signalling. but card don't work. negotiation with E1 fail.
please help me to correct it. i dont' know some parameters such as:
MTU, BAUDRATE
Are you running the voice drivers, or have you configure i
Rich Adamson wrote:
I am new to Asterisk, but so far, having looked at this thread (which is
quite long and full of lot's of hard work), the zttest code, the Digium web
site,
and the spandsp website, and having talked to Digium sales/support about
the TDM, I think the issue belongs the spandsp com
Mike Mueller wrote:
I am new to Asterisk, but so far, having looked at this thread (which is
quite long and full of lot's of hard work), the zttest code, the Digium web site,
and the spandsp website, and having talked to Digium sales/support about
the TDM, I think the issue belongs the spandsp comm
Jonathan wrote:
Andrew Kohlsmith wrote:
BTW are you *really* saving any time by bastardizing your email so
much (ur, u, bcz)... jeez.
I think they teach that crap in school these days ... kids and their
sms cell phones..
I thought it was trying to simulate a high packet loss. :-)
Regards,
Stev
Send an example TIFF file, and I will investigate.
Regards,
Steve
Me wrote:
If the problem is with libtiff, its a problem with every version i've
tried (3.5.7, 3.6.0, 3.6.1, 3.7.1 and 3.7.2)
On 4/30/05, Steve Underwood <[EMAIL PROTECTED]> wrote:
Me wrote:
Hi all,
I'm tryin
Me wrote:
Hi all,
I'm trying to use spandsp and asterisk to send faxes. To do so I am
creating tiffs with Ghostscript. When I use Ghostscript 6.50 it seems
to work fine, but when I create the tiff using Ghostscript 8.51 (or
7.06) txfax garbles the tiff and it comes through all messed up.
First of
Joseph wrote:
Can anybody explain me why IAX is called proprietary protocol?
In some places IAX is refereed as "open protocol".
How can proprietary protocol be open protocol?
Proprietary means it came from a proprietor - Digium in this case. This
is a completely unrelated issue to whether it is
Cyril VELTER wrote:
I just installed it and will keep you informed if a new crash occur, but even
with pre15, crash where not very frequent and usually come in series (~ one
serie of 3/4 crashes every two weeks, so we might have to wait some time...).
I'm pretty happy with the receiving side o
Hi Cyril,
Good work. process_baud is a fairly big routine, and your backtrace
doesn't give the actual line number at which things fall over. However,
studying the code I see that I do not protect against the possibility of
a divide by zero during the initial coarse carrier estimation of any of
Hi Darell,
If FAX works over ulaw over SIP or IAX it is by luck and not by design.
Don't expect it to work, though it often does.
Regards,
Steve
Darrell Collins wrote:
I am having a problem receiving faxes over an IAX connection. If I use
SIP it works fine.
I am running Asterisk 1.0.7. I have a
3% 99.987793%
99.987793% 99.987793%
99.975586% 99.987793%
Thanks
Julian J. M.
On 4/26/05, Steve Underwood <[EMAIL PROTECTED]> wrote:
Why would you expect a bunch of fax modems to work any better than
spandsp? If spandsp doesn't work reliably your system is very likely broken.
I hav
.
Regards,
Steve
Andrew Kohlsmith wrote:
On April 26, 2005 08:12 am, Steve Underwood wrote:
Why would you expect a bunch of fax modems to work any better than
spandsp? If spandsp doesn't work reliably your system is very likely
broken.
I've had spandsp crash out on some kind of floa
Hi Nathaniel,
ETSI ISDN is used by 99% of the world's ISDN E1s, so you can guess the
answer. :-) ETSI ISDN is also known as CTR4, Net5 and most commonly
EuroISDN. It is known as EuroISDN in the * config files.
Regards,
Steve
Nathaniel Angelo A. Torres (247talk) wrote:
Hi, I just wanted to know i
Why would you expect a bunch of fax modems to work any better than
spandsp? If spandsp doesn't work reliably your system is very likely broken.
I have had hundreds of complaints about spandsp reliability. I have
analysed at least 50 or 60 audio logs. I have found maybe 5 or 6 which
has real spa
Rich Adamson wrote:
One way is to buy a relatively inexpensive analog transmission test
set ($400 US). Most have a tone generator and level meter built in.
You didn't mention which country you're located in, but ensure whatever
test set you purchase, that it supports the line impedance in use by
y
Kevin P. Fleming wrote:
Andrew Kohlsmith wrote:
Yes, but then what are you doing with it? You're shuttling the new
data to/from a network card in a lot of cases. Combined with other
traffic over the PCI bus for normal system operation I could see you
coming close to the limitations of regular
Matt Klein wrote:
Kevin,
Mmm. Yep.
-m
On Tue, 12 Apr 2005, Kevin P. Fleming wrote:
Matthew Boehm wrote:
So, no hardware encoding on this beast?
The announcement on the website makes no mention of transcoding, echo
cancellation or toast-and-jam making, so at this time, no, there is
no hardwa
Latency doesn't vary much with the codec. Different codecs package
different amounts of audio per packet, but that is usually a small part
of the overall latency with VoIP.
If G.711 is used with a reasonable quality PLC algorithm, like the new
one in *, it beats G.729 hands down in its packet l
Steve Kann wrote:
Eric Wieling wrote:
[EMAIL PROTECTED] wrote:
Hi,
How can i implement VAD/DTX using zaptel with asterisk towards PSTN.
TDM (PSTN/telcos) do not support VAD. The entire idea of VAD is not
even a valid idea.
Doing VAD on audio coming _from_ the TDM world certainly is something
y
If your hardware isn't getting clean data to spandsp, why should it be
able to get clean data to a hylafax box? Unless you fix the config
problem that stops spandsp working, there is no reason to expect a
pass-through to a modem bank and hylafax to work.
Regards,
Steve
Kevin Brennan wrote:
We a
Asterisk needs more than to just allow T.38 as a codec type. T.38 only
recently gained an RTP option, and very few things support it so far.
You need to have a UDPTL transport for most boxes supporting T.38. I
have a working UDPTL for *, but it needs more polishing before release.
Regards,
Stev
David Hajek wrote:
Hi,
is it possible to use Asterisk with T110P and CAS (channel associated
signalling)?
There are hundreds of CAS protocols. Quite a few currently work with the
T110P.
Regards,
Steve
___
Asterisk-Users mailing list
Asterisk-Users@list
Richard Dutton wrote:
Hi,
I've seen from the Asterisk Hardware list that the Dialogic D/300JCT-1E1 and
D/600JCT-2E1 cards are supported by Asterisk, can anyone tell me if the
D/300SC-1E1 and D/600SC-2E1 cards are as a client has quite a few of these
particular model and would like to use them in an
Dinesh Nair <[EMAIL PROTECTED]> wrote:
On 04/01/05 00:00 Matthew Boehm said the following:
Steve Underwood wrote:
And your EU bias is clearly demonstrated by this. I've never seen a
BRI product outside he EU. :-)
Come to Houston, TX. We were running a BRI for quit
Dinesh Nair wrote:
On 04/01/05 00:00 Matthew Boehm said the following:
Steve Underwood wrote:
And your EU bias is clearly demonstrated by this. I've never seen a
BRI product outside he EU. :-)
Come to Houston, TX. We were running a BRI for quite some time before
upgrading to a T1.
ahem,
Hey Bass,
Tim Bass wrote:
Mr. Underwood,
You might have noticed that I did not start this thread and simply am
agreeing with the original poster.
You might have noticed that I will not be shouted down and insulted to stop
agreeing with the original poster. In fact, if you don't like the thread,
do
You missed:
(4) the server overload caused by people who don't like e-mail lists
telling the people who are perfectly happy with them they are fools.
Wait a moment. I've got it. All these pro-web-forum messages are 1st
April posts, aren't they? :-)
Regards,
Steve
Tim Bass wrote:
The lag time on
Dido Sevilla wrote:
I've been struggling with getting the soft-switch.org Unicall/MFC-R2
implementation working. After wading through some misleading
instructions and a build process that leaves much to be desired in
terms of reliability and consistency (e.g., the order in which the
various librari
Chuck Bunn wrote:
Hi,
I am new to Asterisk and the first thing I have noticed about Asterisk
and Pingtels open PBX's is that they are using this dinosaur method of
running forums. It is a real pain getting every message in the forum
and essentially keeping my own database of issues. With that sa
Remco Barende wrote:
It would be nice if Digium would accept the bristuff patch at some
stage and include it in asterisk.
GPL code cannot go into the Asterisk distribution.
Regards,
Steve
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
ht
Eric Bishop wrote:
True. I think Digium's USA bias is clearly demonstrated by their lack
of a BRI ISDN product. Most of the rest of the world use it in
abudnace yet Digium do not see fit to service this market because it
is not big in the US. very poor...
And your EU bias is clearly demonstr
Hi,
You can write a GPL'ed SS7. There is nothing protected in the SS7
design. I don't think there ever were any patents. However, if there
were they ran out long ago.
Our non-GPL SS7 (because it is commercial) stack is written as a library
in C. A modified chan_zap links it into Asterisk at the
Gustavo García wrote:
Hi everybody,
GIPS have different products, not only codecs:
* Voice enhancements: packet loss concealment algorithms, noise concealment,
jitter buffer, agc, aec (can be used with any codec)
* Codecs: iLbc (free), ISAC, G711 Wideband...
You can include in asterisk voice e
501 - 600 of 1161 matches
Mail list logo
