Hi,
On the voip-info.org page for Asterisk FAX, someone has added a note
saying Pika have stopped selling their soft FAX add on for Asterisk. Can
any confirm or deny this? I found it strange, as it appears Pika have
recently licence a V.34 FAX modem from Commetrex. I assumed they were
going
Hi,
I would be interested in any reports of anyone getting a T.38 FAX to
send or receive successfully with Zoiper. I've tried to test my T.38
implementation against more than one revision of Zoiper, and I yet to
see it behave sanely.
Steve
Olivier wrote:
2008/12/3 [EMAIL PROTECTED]
Jean-Denis Girard wrote:
The price of RealSpeak is not far from an order of magnitude higher
compared to Cepstral.
Only an order of magnitude? They've reduced it a lot then. :-)
Steve
___
-- Bandwidth and Colocation Provided by
Olivier wrote:
Thanks for this detailed reply.
I was trying to test SendFAX, ReceiveFAX as first on my way to Hylafax
with either iaxmodem or t38modem.
Have you tried any of those 2 (iaxmodem or t38modem) ?
Which one would you pick ?
iaxmodem only does audio FAXing (for the present).
Benny Amorsen wrote:
Steve Underwood [EMAIL PROTECTED] writes:
In spandsp I do the G.711 conversions algorithmically. Most modern
processors have a where is the top 1 instruction, and that reduces the
calculations to something very fast.
Very nice! I'd like to see the code
Benny Amorsen wrote:
Interestingly the Linux kernel has this for find-last-set:
static inline unsigned long __fls(unsigned long word)
{
asm(bsr %1,%0
:=r (word)
:rm (word));
return word;
}
spandsp has this (Everything non-x86 has been removed):
Matthew Fredrickson wrote:
Actually, with the way caching is done on nearly all modern processors,
it is debatable whether or not a look up table is the optimal way to do
the conversion, at least on such a simple codec such as ulaw or alaw.
In fact, the amount of time it takes to fetch
Wilton Helm wrote:
I'm a bit puzzled, also, having implemented ulaw and alaw in an
embedded application. Each can be done with a 16 Kbyte table in about
0 time with no errors. There are probably tricks that will cut the
table down by 2 or 4 X for a small cost in CPU cycles. The inverse
Jonn R Taylor wrote:
I have been able to repeat the results at other locations. The location that
has 26 pages is a linksys PAP2T our accounting person uses remotely to fax
stuff to the office. The ATA is behind a DIL-625 router with QOS on a DSL
line.
I can send faxes from my test sever
Hi,
A lot of people talk about grooming to make VoIP work smoothly, not just
for FAX. However, most people can only achieve grooming in one
direction. Their ISP will not cooperate, and groom what is sent to the
subscriber. Unless you just keep your DSL link very lightly loaded, by
doing no
JD wrote:
Gordon Henderson wrote:
but it's very do-able, given good Internet connections.
[...]
I think your statements were just a bit too strong - I agree
wholeheartedly about the V. protocols and copper, but I've found in
practice that faxing over IP is not just
Olivier wrote:
2008/10/24 Wilton Helm [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
I've been following this thread and trying to sort out what is
wanted, what is available, and why. Comments to the following
would be appreciated and might be useful to others.
1.
Benny Amorsen wrote:
Steve Underwood [EMAIL PROTECTED] writes:
That list rather poorly supports your argument. The PAP2 and the PAP2T
do *not* support T.38, despite numerous arguments you'll find to the
contrary. Personally I believe Linksys, the manual, and the menus
Benny Amorsen wrote:
Steve Underwood [EMAIL PROTECTED] writes:
Even the big floor standing office MFPs typically only offer T.37 or
T.38 only through an expensive option card.
Medium MFP's almost all support T.37. They call it scan to email,
but they do it (as far as I can tell
Olivier wrote:
Linksys PAP2 or 3102 for instance
or Patton M-ATA
In fact, I would say most analog gateways with FXS port should also
support T.38.
In this case, your setup would be :
That list rather poorly supports your argument. The PAP2 and the PAP2T
do *not* support T.38, despite
Hi Peter,
Thailand is similar to China, except for two things.
- Some places require that billing pulses be generated.
- There may be places using DTMF instead of MFC.
The first issue is definitely the case. The second might just be the
false reporting of issues. People definitely used
Atis Lezdins wrote:
On Wed, Oct 1, 2008 at 6:34 AM, Andrew Joakimsen [EMAIL PROTECTED] wrote:
On Sun, Mar 23, 2008 at 11:34 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
It is completely illegal in any country that recognizes patents.
You mean countries that recognize
Atis Lezdins wrote:
On Wed, Oct 1, 2008 at 10:12 AM, Steve Underwood [EMAIL PROTECTED] wrote:
Atis Lezdins wrote:
On Wed, Oct 1, 2008 at 6:34 AM, Andrew Joakimsen [EMAIL PROTECTED] wrote:
On Sun, Mar 23, 2008 at 11:34 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote
Michael Graves wrote:
Hi All,
So I've been exploring the use of G.722 encoded wideband audio
recently. I have three different SIP devices that allow this: Eyebeam,
IP650 and a Siemens S865IP. The Siemens and IP650 seems to work fine
together. Calls pass between them in what the Polycom notes
Michael Graves wrote:
On Thu, 4 Sep 2008 08:48:47 -0500, Russell Bryant wrote:
Asterisk should work fine with any phone that supports that codec.
Personally, I have only used it with Polycom phones. Also, again,
Asterisk 1.4 only supports G722 passthrough, where as Asterisk 1.6 has
Hi John,
In China you will generally get either MFC/R2 or EuroISDN. MFC/R2 is
much more common.
The only oddity with EuroISDN is that it often provided without CRC4.
That doesn't make a lot of sense, but there it is. MFC/R2 seems to be
universally provided without CRC4 in China.
You said you
Kevin P. Fleming wrote:
Olivier wrote:
I thought HPEC was licenced by Digium from Octasic (ie those 2 software
are the same).
Maybe someone should correct me ...
That is not correct; HPEC is a G.168 line echo canceller from Adaptive
Digital Technologies. The same algorithm (but
Kevin P. Fleming wrote:
Gordon Henderson wrote:
So at worst, it's saying it can handle 29 incarnations, and at best, 37 -
that's assuming no other CPU load such as transcoding.
So it's well capable of handing your requirements of 16 channels - more-so
if you're using a server class
Kevin P. Fleming wrote:
Steve Underwood wrote:
You might think a standard phone plugged into an adaptor, like a
Magic-jack, would be limited to narrow band voice, as that is all the
phone was designed for. It turns out most phones only aggressively
filter at the low end of the band
Matt Watson wrote:
I believe HPEC actually is the same algorithm (G.168) that the HW echo
cancel modules use.. the difference being that HPEC uses up CPU cycles
and its performance will be impacted on a system with higher CPU load,
whereas the HW modules have a dedicated DSP for it.
G.168
Dean Collins wrote:
1/ RD costs v's number of units manafactured per annum.
The only business phone I ever contributed to had a run rate of about
500K/annum and a production life of multiple years (not sure how many it
lasted for). I put some (at the time) exotic DSP into a high end
Anthony Francis wrote:
Steve Underwood wrote:
C. Savinovich wrote:
I am puzzled by the quality of magicjack. I keep trying to figure out how
they can the quality be that adequate. Since Skype also has an excellent
quality, that leaves me to believe that software based calls
Tzafrir Cohen wrote:
On Sat, Jul 12, 2008 at 10:26:24AM +0800, Steve Underwood wrote:
C. Savinovich wrote:
I am puzzled by the quality of magicjack. I keep trying to figure out how
they can the quality be that adequate. Since Skype also has an excellent
quality, that leaves me
Dave Cotton wrote:
Joseph wrote:
On 07/11/08 18:37, Dave Cotton wrote:
SIP wrote:
Joseph wrote:
I need another Sipura 3K and the replacement I think is Linksys SPA3102.
Any input on how reliable is it?
We have a few dozen subscribers using them
C. Savinovich wrote:
I am puzzled by the quality of magicjack. I keep trying to figure out how
they can the quality be that adequate. Since Skype also has an excellent
quality, that leaves me to believe that software based calls (softphones)
could have and advantage over hardphones, provided
marek cervenka wrote:
hi,
there is T.38 fax gateway for asterisk
http://bugs.digium.com/view.php?id=12931
please test it and report bugs
for people from
http://www.voip-info.org/wiki-Asterisk+T.38+Bounty
if you still want donate t.38 development please contact me at cervajs at
List - Non-Commercial
Discussion[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED].com
Steve Underwood wrote:
marek cervenka wrote:
hi,
there is T.38 fax gateway for asterisk
http://bugs.digium.com/view.php?id=12931
please test it and report
Vinícius Fontes wrote:
When people release software under the GPL license, like Steve Underwood did
with libunicall, spandsp and so on, they were supposed to know that other
people has the right to use their code.
The problem is that almost any licence term which tries to limit
Mariano Borgognone wrote:
Moises, we've already set debug level at 255 on unicall.conf and at
logger.conf we've enabled full log (notice,warning,error,debug,verbose).
Has anyone experienced with a Siemens EWSD switch?
Anyone knows about to change R2 timers at unicall.conf ?
Please any
Michael Graves wrote:
Which flavor of G.722 has been implemented in Asterisk? And starting
with what release version?
The only flavour with a defined RTP format is the full 64kbps one.
Steve
___
-- Bandwidth and Colocation Provided by
Matt Watson wrote:
I believe Asterisk 1.6 with app_fax supports T.38 origination and
termination, that is not gatewaying, however if origination and termination
are already there, gatewaying should be fairly trivial to implement. I
haven't actually tested 1.6 using T.38, however I have
Andrew Kohlsmith (lists) wrote:
On April 7, 2008 02:01:08 am Alex Balashov wrote:
A Lucent TNT Max outfitted with _plethoric_ VFCs might work okay. Apex
too, perhaps. Haven't tried to see how much it can handle when TDM-RTP
translation is required.
I'm curious; are the
Mindaugas Kezys wrote:
Hello,
Higher speeds then 9600kbps are not permited by patents.
Would you care to name one that prevents 14,400?
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO - Advanced Billing Solution for Asterisk PBX
-Original Message-
From: [EMAIL
Ricardo Carvalho wrote:
I made some tests with FAX in Asterisk 1.4 using T.38 between two ATAs
connected to legacy FAX machines, and realized that only SIP can make
passthrough in the server while RTP go direct between endpoints. Is it
possible for RTP data stream also to make passthrough
Benny Amorsen wrote:
Steve Underwood [EMAIL PROTECTED] writes:
Try reading the GPL and the FSF's interpretation of it. If things are
running in the same address space as my code, they need to be GPL
compatible, or I am likely to take action.
The GPL is not an EULA. You don't
,
Fernando
Thomas Kenyon wrote:
Steve Underwood wrote:
I thought * was still not capable for T.38 gateway operation. Doesn't
beta 4 just added T.38 termination? And, I believe it misses out some
key elements of doing that properly. Note that T.38
Rob Hillis wrote:
T.38 is for all intents and purposes a codec. It's purpose is to
re-encode a fax transmission as a data stream to be re-assembled at
the other end as if it were a fax call. Seems to me to be pretty
close to the definition of a codec to me.
T.38 is not a simple
gateway.
Steve Underwood wrote:
Rob Hillis wrote:
Not unless you're running CallWeaver or Asterisk 1.6.0-beta4. Asterisk
has had passthrough support for T.38 for a while (somewhere in 1.4 it
became available IIRC) but is currently completely incapable of
terminating or encoding a fax call
Zeeshan Zakaria wrote:
Hello everyone,
Some months ago there were news about J2 filing lawsuits against
companies using fax-to-email technology, as they claimed it was their
patent. They had also won some cases, until someone filed a counter
lawsuit against them based some other grounds
Rob Hillis wrote:
Not unless you're running CallWeaver or Asterisk 1.6.0-beta4. Asterisk
has had passthrough support for T.38 for a while (somewhere in 1.4 it
became available IIRC) but is currently completely incapable of
terminating or encoding a fax call to T.38.
I thought * was
Hi Rob,
Rob Hillis wrote:
Well that answers that question. I see that t38modem provides an H232
modem - is this unsuitable for HylaFAX's purpose? (ignoring the fact
that it requires a kernel recompile on most newer distros.)
Steve Underwood wrote:
Rob Hillis wrote:
Last time I heard
Jonn R Taylor wrote:
I have always said that if some one said it can't be done, they did not try
hard enough.
FYI... I love this.
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
As the person
Rob Hillis wrote:
Last time I heard IAXModem didn't support T.38 because the IAX2
protocol didn't support T.38 - whether that's still the case or not, I
don't know.
There are actually two reasons. One is that T.38 over IAX is not
defined. The other is the current T.38 termination support in
Roger C. Beraldi Martins wrote:
Moises,
I try put the line exactly like you send me, saw the time wait getting
longer with the parameter you describe to increment. But the error is
the same as you can see in logs.
Has other way to solve this problem, may I question to my telephony
Hi Josué,
Those E/F mismatch issues are due to using incompatible versions of
spandsp and unicall.
MFC/R2 defines 15 tone signals. These are called 1 to 15 in the R2
documentation. I wanted a single character code for these, so I used 0-9
for the digits, and A-E for the other 5 codes. This
Darryl Dunkin wrote:
Does anyone have any opinions on the music on hold quality over G729?
The stock files seem to sound terrible over it, this is enhanced further
by calls coming from the PSTN via a Zaptel gateway. I am only using the
stock wav files and have not attempted to use much else so
Michael J. Liberatore wrote:
Alex, I thought asterisk 1.4 supports faxing internally now without the
need for extra software? Is your solution a different one? I have no
experience with faxing yet but plan to soon, that's why I ask and will
read your blog entry.
You need extra software,
Jakub Syrek wrote:
All errors was genereted by physical link.
Protocolvariant cz,10,6 its ok for me in Poland
Thanks for help
Regards
Akron
Thanks. I will make a note of that in the code.
Steve
___
--Bandwidth and Colocation Provided by
Hi Jakub,
Most countries which used to be part of the iron curtain block, back in
the good old days, use the same protocol. Try the Czech variant. It will
probably be OK for you. If it works, please report that, and Poland can
be added to the list of variants.
Steve
Jakub Syrek wrote:
Im
Steve Totaro wrote:
Tzafrir Cohen wrote:
On Tue, Oct 30, 2007 at 04:49:25AM -0700, satish patel wrote:
Dear all
This is survey of Digium Vs Sangoma Hardware i am
going to purchase some Asterisk supported hardware and i have
confusion between both company
Steve Davies wrote:
On 8/29/07, Christian Peter [EMAIL PROTECTED] wrote:
Am Mittwoch, den 29.08.2007, 20:13 +0800 schrieb Steve Underwood:
Christian Peter wrote:
Hi list,
I'm running current SpanDSP
http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre6.tgz
Alan Lord wrote:
Steve Davies wrote:
On 10/30/07, Steve Underwood [EMAIL PROTECTED] wrote:
This was fixed a few weeks ago. There was an error in the FAX decoder,
but only a very few encoders create images that hit the issue. If you
try 0.0.4pre11 you will find it fixes several other
Smith, Rick wrote:
doesn't look legit to me.
It's got CE/FCC emblems, but no ID #'s ?!
If that is a mark of legitimacy, then most equipment must be fake. :-)
Steve
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
Olivier wrote:
I was told yesterday (by Cantata guy) that T.38 demands a good level
of QoS.
That surprised me a lot as I thought the whole purpose of T.38 was to
avoid SIP and ToIP latency.
T.37 is the answer to reliability, but most people don't want to use it
for totally stupid reasons.
Brian West wrote:
I would like to point out that G.722 is a really awesome codec for
wideband. Asterisk has some changes that will need to be made to
support variable audio rates. We did this in FreeSWITCH from the
start. I think Asterisk will be doing similar things to bridge an 8k
to
successfully.
Shouldn't you have used spandsp 0.0.3 with asterisk 1.2 ?
Actually, as Steve Underwood has gently reminded the list several
times, he recommends SpanDsp 0.0.2 for Asterisk 1.2
Well, its not so much that I recommend it. Its just that I have never
done anything
Christian Peter wrote:
Hi list,
I'm running current SpanDSP
http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre6.tgz
with Asterisk 1.2.22 somewhat successfully.
Most Fax machines do work but I have problems with people having
Tobit FaxWare and Shamrock CapiFax.
shadowym wrote:
Well there are a couple fine examples of FUD if I do say so myself. Just do
a search and see what cards the 'serious' companies out there are using.
Nuff said.
When did saying nothing at all become enough?
Eric ManxPower Wieling wrote:
Sangoma cards are complicated to
Zeeshan Zakaria wrote:
What exactly is patented by J2? Is it receiving fax over the Internet,
converting to PDF and sending as an email attachment using sendmail or
postfix etc? Or is it receiving it through PRI, or PSTN line over the
computer and converting and emailing? What exactly they
Henry L.Coleman wrote:
This legal question pops up every now and then, and depending on how
paranoid you are you can eventually start thinking that the US patent
office is under your bed.(I'm just checking now)
First thing to note is that you aren't worth suing. This is a game that
only
Patrick wrote:
Hi all,
Anyone have an idea which version of spandsp, libunicall, libmfcr2,
libsupertone, app_rxfax/app_txfax and chan_unicall I should use for the
latest asterisk 1.2?
Would that be the ones listed below?
http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre4.tgz
Moises Silva wrote:
The latest versions of unicall (0.0.5) work with the latest spandsp
(0.0.4), but I have done nothing about making either of them work with
Asterisk.
Minor changes were needed to chan_unicall. Anyone interested in using
it can find it here:
Moises Silva wrote:
I would not call that properly a fix. We need to know why is failing
in newer spandsp versions in the first place. Can you make a diff and
post it?
Why are people so determined to break things. If you want to use
unicall-0.0.3pre11, use it with spandsp-0.0.2.
The
Carlos Chavez wrote:
It seems the problem with Unicall and Nextel is also present in
Asterisk 1.2 and not only in 1.4. I decided to downgrade from 1.4.9 to
1.2.23 so the customer could have CID and calls from Nextel but today he
told me that they cannot receive any calls from Nextel,
Victor Toofic wrote:
El Sun, Jul 29 de 2007 a las 20:04 +0800, Steve Underwood comentaba:
What versions of software did you use to get a screwed up result like
that? The message Don't know how to handle signalling event Accepted
is printed at the end of a case statement which does handle
Hi Victor,
Victor Toofic wrote:
Hi,
I've finally got running Asterisk 1.2.14 with UniCall MFC/R2 patches, I
can generate calls and all seems OK but I cannot receive any call, this is
what I get:
Unicall/3 event Offered
CRN 32769 - Offered on channel 0 (ANI: 814777, DNIS: 83329276,
Alvaro Parres wrote:
Search at mfcr2.c this:
case MFCR2_PROT_MEXICO:
And add the next line after that line:
mfcr2-group_i_end_of_ANI_restricted = R2_SIGI_12;
This will help you on calls that have the restricted flag on the ANI
only. (Nextel). But not on no caller id calls.
Carlos Chavez wrote:
On Tue, 2007-07-03 at 23:50 -0400, C F wrote:
Change it to ISDN. There is no point in not to, what card do you have
in the TDA200? A PRI or or just T/E1? Since it's too differenct cards
on the TDA200. In fact accroding to Panasonic CallerID isn't supported
on none
Dean Collins wrote:
Nuance Communications has agreed to buy Tegic Communications, the
developer of the T9 predictive text input software for mobile phones,
from AOL for $265 million in cash.
http://www.wirelessweek.com/article.aspx?id=149702
Article goes on to say T9 is in use on over
Gary Chen wrote:
Does anybody know why Asterisk does not support inband DTMF for G.729?
Our SIP carrier use inband dtmf for G.729. This causes problem for us
to use it for our Asterisk IVR system.
Any suggestion to solve this problem?
I supposed the basic why is nobody has done it.
G.729
Hi all,
If anyone has access to the manuals for a TAS240 voice band subscriber loop
emulator or a TAS
Series 2 Digital network emulator, I'd love to here from you.
Thanks.
Steve
___
--Bandwidth and Colocation Provided by
Paradise Dove wrote:
can anybody help me to choose the most reliable fax solution for * .
after googling the net i found that there are at least two solutions
for this, app_rxfax+spandsp and iaxmodem+hylafax.
- what's the differences between these two?
- which one's better? why?
Peter Gubis wrote:
Hi,
connection is already established (i can also hear called person for a
while). Problem is, that the line is every time dropped after 1 second.
I assume, that first billing pulse arrives immediately after link is
established and it drops the line.
Next week I'll be
Stephen Davies wrote:
On 09/06/07, Deepak Naidu [EMAIL PROTECTED] wrote:
Ya, I have done that, below is zapata.conf. Also we had an TMP card
with
analog lines. SIP cals were great on them. now when we switched
over.
SIP calls have echo.. which shouldnt be at all.
If you are getting echo
Steve Totaro wrote:
If I am not mistaking, g711u is ulaw. Ulaw and Alaw are the best
since they are lossless, meaning no
Lossless? Our friends at http://en.wikipedia.org/wiki/Ulaw wouldn't lie. :-)
Steve
___
--Bandwidth and Colocation provided by
Alain Degreffe wrote:
Why do you use Ulaw as codec ?
Try another codec ( g729 is by far the best but isn't free ).
G.729 isn't the best. Its just the one you need to be compatible with
the other end. G.729 is the lock-in choice, not the quality choice.
Steve
Dovid B wrote:
snip
ROTFL. The US patent system is treated with contempt in Hong Kong?
You have no idea how EXTREME legislation in Hong Kong against IP
'theft' is in Hong Kong.
/snip
I find this hard to believe since most hack attempts to my box's
originate from IP's in China.
What exactly
Andrew Joakimsen wrote:
I'll consider the offer if it includes your code being included with
Asterisk.
On 3/3/07, Steve Underwood [EMAIL PROTECTED] wrote:
I'll do it for 30% less than they quote. :-)
So, you want a discounted price for something that offers more than
Attractel will offer you
Zoa wrote:
So does asterisk (Albeit with a commercial package)
http://www.attractel.com/t38.html
I'll do it for 30% less than they quote. :-)
Lee Howard wrote:
Matt Riddell [NZ] wrote:
Does OpenPBX do a T.38 gateway then?
Yes, it does.
Lee.
Steve
Zoilo Gomez wrote:
I really don't get it
From several emails in this list archive, I had clearly understood
that it is important to switch Echo Cancellation off for fax-channels,
or faxing would not work properly.
However, faxing (B410P ISDN bridged to TE410P PSTN) seems to work fine
Wireless wrote:
- Original Message -
From: Nic Bellamy [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, February 13, 2007 8:40 PM
Subject: Re: [asterisk-users] The High Performance Echo Canceller (HPEC)
Andrew Kohlsmith wrote:
On Thursday 15 February 2007 6:51 am, Steve Underwood wrote:
It looks like octasic have started supplying their echo canceller as
host software for zaptel now. I expect either canceller would work with
the Sangoma cards, as they currently sit in the zaptel framework
Dean Collins wrote:
How do you fake echo for testing purposes then?
I you look in my spandsp library, thre is a fairly complete framework
for testing echo cancellers according to G.168. This includes modelling
the various echo patterns defined in G.168.
Steve
Tzafrir Cohen wrote:
On Thu, Feb 15, 2007 at 05:35:09PM -0500, Dean Collins wrote:
How do you fake echo for testing purposes then?
A soft phone will normally give you good enough delay. Call a an analog
phone from a different soft phone. Make sure no over-freindly
device/software
shadowym wrote:
The algorithms may be similar but EC is an infinitely variable
non-linear(analog) process. A CPU cannot do that. You can fake it by
performing cpu intensive rapid calculations one after another but it is
fundamentally not an analog processor. HWEC is designed to deal with the
no interest in any further interest in
expanding T.38 or faxing support in Asterisk.
Steve Underwood and the other fine persons that have helped to develop
the software DSPs and other stuff required for FoIP support also have
no interest in writing any further faxing support for Asterisk (RxFax,
TxFax
Hi,
Gordon Henderson wrote:
On Sat, 20 Jan 2007, Andrew Joakimsen wrote:
Assuming your PRI supports timing from the remote end (CO) which I
highly suspect is the case, then you should set the asterisk machine
to be a slave to the CO timing and then set any other interfaces you
have to NOT be
Leo Ann Boon wrote:
Andrew Joakimsen wrote:
Most of the Cisco phones sold cheap are UNLICENSED (global spare)
thus you would not be able to purchase (or at least aren't supposed
to) the smartnet contracts, you need to buy the license ($100+) and
the contract ($10 or so)
I'm always surprised by
Paul Hales wrote:
When you built Asterisk, it must have refused to build the ilbc codec -
I have never seen an Asterisk box that could not transcode ilbc, in over
3 years of working with Asterisk.
PaulH
iLBC is not compatible with GPL code. So, if you are using GPL code,
such as MySQL or
Matthew Fredrickson wrote:
On Nov 3, 2006, at 9:30 AM, Olivier wrote:
Hi,
From a chat with Kevin Fleming during last Astricon Europe in Paris,
I understood that Digium was about to support TDM bridging with
TDM2400 board.
This feature had to be introduced with 1.4 release.
Is my
Leo Ann Boon wrote:
Gregory Duchatelet wrote:
Hi all,
Does Asterisk now support Intel’s HMP platforms ? Does it support in
1.4 version ?
There's a special driver for Intel-based HMP hardware+software for
ABE. On the other hand, Asterisk has always been doing HMP :). In
fact, I would
Andrew Joakimsen wrote:
But if we have asterisk and add on Steve's code wouldn't it (suppor to
recieve a t.38 fax call and have spandsp decode it) work? What does
Steve granting a license to Digium have to do with it? I don't care if
Asterisk and the fax support don't come from the same
Andrew Joakimsen wrote:
You are using bad software to view the faxes. In Windows the picture
and fax viewer seems to work fine, however in Linux KGhostView or
whever the default program is does not work, however you should try
KFaxView.
They've broken Windows now, if you install Microsoft
M. Shokuie Nia wrote:
Dear folk,
My problem solved after two day research and try and error method ;). It was
related to rxgain of the board im using. I've set the rxgain to 12 and it
seems made some problem. As far as I got the spandsp is so sensitive about
noise on the line and because of
Giedrius Augys wrote:
Hi,
Now I'm testing faxes with spandsp. I have problems that spandsp do
not add headers to fax page: LOCALHEADERINFO.
Please help me.
There is a bug in adding page header with spandsp-0.0.2pre26. I have
fixed this in the development code, but I haven't yet put the
201 - 300 of 1116 matches
Mail list logo
