Hi Terry,
People are doing this kind of thing, but I am not. I don't know quite
how they do it. Hopefully someone will pipe in with a wonderful solution :-)
Regards,
Steve
Terry Goodwin wrote:
Damn! :-(
Now that I have spandsp working on my * I was going to try and get it
working with hylafax.
Scott Stingel wrote:
Not to mention that the Digium T1/E1 cards are about 1/5 the cost of
Dialogic's. That said, it should be noted that the Dialogic cards allow
scaling to a much larger size within one chassis, thanks to heavy DSP
horsepower helping out with the low level chores.
The Dialogic
Jason Williams wrote:
At 09:16 27/05/2004 -0500, you wrote:
Maybe the time and effort would be better spent finding out why the
Digium card won't work on the NTL's PRI and either fixing it or
providing the information and testing facility to someone who can.
NTL's PRI uses ISDN 85 not q931 so a
PROTECTED] On Behalf Of Steve Underwood
Sent: Thursday, May 27, 2004 7:48 AM
To: [EMAIL PROTECTED]
Subject: Re: dialogic was RE: [Asterisk-Users] Glare condition - How well
does asteriskhandle?
Scott Stingel wrote:
Not to mention that the Digium T1/E1 cards are about 1/5 the cost of
Dialogic's
tim panton wrote:
Steve Underwood wrote:
Jason Williams wrote:
At 09:16 27/05/2004 -0500, you wrote:
Maybe the time and effort would be better spent finding out why the
Digium card won't work on the NTL's PRI and either fixing it or
providing the information and testing facility to someone who can
Scott Stingel wrote:
Hi Steve-
Just briefly:
I was mentioning the old days to illustrate what an even low clock rate DSP
can do. More recently (2000-2001), using D/600's we were able to drive a
large number of channels (8-12 E1's) for IVR.
Ah, the D/600 - damned big heat sink, lots of heat,
Christopher Lewis wrote:
I appreciate all the support SpanDSP is getting on this list, and the work
that went into developing it. It's nice to see that Asterisk actually
supports sending and receiving faxes now.
One thing I was expecting to see here is that HylaFax works with SpanDSP, but
I
Hi Dan,
Any just as with those others, the cure is the same. Don't use libtiff
3.6.1. Its fax image support is broken.
Regards,
Steve
Dan Cunningham wrote:
Like some others on the list spandsp is segfaulting asterisk when recieving
a fax. I'm on debian testing/unstable with freshly checked out
Hi Terry,
You could have tried following the instructions at
fttp://www.opencall.org/instructions.html rather than guess how to build
the software. :-)
If you are using the latest CVS there is a build problem with app_rxfax
and app_txfax. If you use the latest *stable* CVS it should build OK.
Rich Adamson wrote:
Using the fxs_ls settings, the TE410P does not see
when the remote side drops the call.
Is em signaling a more reliable way to go?
em signaling was designed years ago to address the problems that
existed with loop start, so yes it is more reliable.
Eh? EM predates loop
Scott Stingel wrote:
A little more research on this:
I found a Dialogic flow diagram that seems to indicate what happens when
glare occurs on an IDSN line. So it looks like perhaps it can occur?
Re-phrasing my original question: Does the asterisk PRI driver properly
re-try an outgoing call that
an obvious way to detect./observe these events?
-Darren
--
Darren Nickerson
Senior Sales Support Engineer
iFax Solutions, Inc. www.ifax.com
[EMAIL PROTECTED]
+1.215.438.4638
+1.215.243.8335 (fax)
- Original Message -
From: Steve Underwood [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, May
the frame slips you say we all have' ;-)
-d
--
Darren Nickerson
Senior Sales Support Engineer
iFax Solutions, Inc. www.ifax.com
[EMAIL PROTECTED]
+1.215.438.4638
+1.215.243.8335 (fax)
- Original Message -
From: Steve Underwood [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, May 24, 2004
://www.psknet.com
866.477.5638
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steve Underwood
Sent: Saturday, May 22, 2004 10:27 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] RxFAX generates no tiff file
Hi Troy,
People had a lot of problems like
receiving 1/3 pages from faxis
?
who is a problems-)
I
Steve Underwood wrote:
Hi Troy,
People had a lot of problems like this with earlier versions of
spandsp. However, the latest version is pretty solid, and people are
using it in high volume production applications. If you are getting
these bad results
Hi Mike,
Your log seems to be incomplete. It stops in the middle of the call.
Regards,
Steve
Mike Heininger wrote:
Hi,
I am trying to receive a fax with the spandsp library.
The sending fax says success but there is no tiff file generated.
I use exten = 7000,1,rxfax(/tmp/testfax.tif) in my
Hi Troy,
People had a lot of problems like this with earlier versions of spandsp.
However, the latest version is pretty solid, and people are using it in
high volume production applications. If you are getting these bad
results with the latest version I would be interested to see the audio
log
Hi Mike,
How do you run rxfax? You problem is probably something to do with that.
Your's is the first report I have had of no TIFF file whatsoever.
Regards,
Steve
Mike Heininger wrote:
Am 22.05.2004 um 20:09 schrieb Troy Settle:
Dunno about not being able to generate a tiff, I got rxfax to do
Sergio Serrano Revuelto wrote:
Gsm is wav in 8/mono
srsergio
Oh, no it isn't
Regards,
Steve
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[EMAIL PROTECTED] wrote:
Folks seem to have forgotten that
the original hackers were hacking
stable and secure traditional PBXs
with captain crunch whistles!
Mitnik ran wild through PBX's and mobille networks.
Let's work to set up secure VOIP, but
don't let anyone kid you about the golden
on
what I can do to fix this problem?
Looking forward to your responses.
Regards,
Jorge
On Fri, 2004-04-30 at 22:58, Steve Underwood wrote:
jorge verastegui wrote:
Hi
i have successfully downloaded and compiled libr2 from source.
But i dont seem to find how to properly configure
Peter Corlett wrote:
My employer wants to use Asterisk, but the E1 circuit providing the current
phone system is DASS2 rather than ISDN30. Converting the E1 to ISDN30 is not
a practical proposition at this time as it'd stop the legacy phone system
from working.
Is there any sort of hardware
Steven Critchfield wrote:
On Thu, 2004-05-13 at 12:07, Andrew Kohlsmith wrote:
17 years for software patents is FAR too long, IMO, but that's an entirely
different story. IMO software patents shoudln't be for more than ~24 months
since the industry moves so blazingly fast.
I'm of
Chinese DVD player makers can ship a player for $25. Of that, they pay
up to $5 in IP royalties to foreign companies. The development of their
own EVD standard was specifically to sidestep this burden. Patents do
cause harmful splintering. We need stuff that interworks, far more than
we need
Hi,
DTMF will not pass reliably through *any* low bit rate codec, with the
exception of some forms of ADPCM. GSM certainly does not work. When a
VoIP system is configured properly it works just like a cellular
telephone system, which suffers the same problem. It doesn't send the
DTMF tones.
brian k. west wrote:
nasty. The higher rate (6.something kbps) sounds more reasonable. Using
30ms blocks, it is not so compatible with *, which is geared to 20ms
block processing. A lost packet causes a 30ms hole, so it tends to be
less tolerant of packet loss than something working in smaller
Jeremy McNamara wrote:
Togan Muftuoglu wrote:
and what will happen if the box has more than one ethernet card
Mark is smarter than Voiceagehe will make it work.
Jeremy McNamara
That isn't saying much. The village idiot is smarter than VoiceAge. :-)
Regards,
Steve
Craig wrote:
Greetings all,
I have searched all over and have found bits and pieces on low bit rate
codecs, however I have found it very difficult to compare apples with
apples.
The conclusions I have come to are as follows, I would appreciate if
anyone has some feedback, or point me to where I
Bartosz Jozwiak wrote:
Hi All:
I'm a newbee to Asterisk. I currently working on a project and want to
know
if Asterisk does support R2 Signaling.
Thanks
Begra8fl
Yes I think so. But you have to download libr2 and compile it, if I am not
mistaken.
You are mistaken.
Regards,
This is not a duplex card. It won't work with *
Regards,
Steve
Alejandro Acosta wrote:
Hello,
I just wanted to know if any of you has successfully (or know about)
installed the Dialogic 4xFXs ISA CARD D/41 ESC? Does it work with *?,
if so, how is the driver called?
Thanks a lot for your
Most Canon models seem to be working OK. Serge's Canon drops the line
every time during negotiation, as though it has decided spandsp doesn't
support compatible capabilities. It is not clear why this is happening,
though. The modes which spandsp is asking to use are the most widely
used ones.
Technology, Inc.
Palo Alto California London England
www.evtmedia.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Underwood
Sent: Saturday, April 24, 2004 10:20 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] D/41 ESC dialogic ISA CARD
If you used the default build and install configuration it will install
the library in /usr/local/lib. Is that in yout library paths?
Regards,
Steve
[EMAIL PROTECTED] wrote:
I successfully compiled installed the
spandsp-0.0.1k.tar.gz modules for faxing and
patched the asterisk
Andrew Kohlsmith wrote:
[...]
Not at all. Any of the channel banks I've tested have better echo and audio
quality than the X100P. I believe it comes down to the Part68 interface
being better able to accomodate different lines but YMMV. I have never had
decent results with an X100P. All of
Andrew Kohlsmith wrote:
When we had a MCI ct1, they couldn't send us proper supervised hangup on
a loopstart encoded DS0. They claimed it to be a problem with the
software on their switch. Their solution was to switch to groundstart.
Our end solution was to drop them and switch to
Mike Machado wrote:
cvs HEAD did infact fix the ringing problem. Thanks Eric!
I have another question for all you T1 buffs out there. The T1 I am
working with goes into our local phone switch (Excel switch). Currently
we are using E M Wink signaling. The problem is we cannot set callerid
on the
Some people have some really wacky ideas about how sampled systems work :-)
Regards,
Steve
Michael Welter wrote:
Just when I thought I couldn't be wrong, I was wrong. We have
woodpeckers that drill into the arial telephone cables, and water
seeps through the holes and partially grounds the
Stephen Davies wrote:
Hi Alex,
Indeed the call end termination doesn't work on an NTL line. I'm not
so sure it works too well on other lines either.
I did some work a while back to add detection of the UK busy/hangup
signal on the line, but I never got it working well enough to depend
on it.
Duane wrote:
Adam Goryachev wrote:
I am in Australia, which I think expects callerid at a different time to
other countries Although other people have told me callerid is
working correctly for them
Forgot to mention there is a patch for this, but it won't patch
cleanly against current
Hi Michael,
I tried to reply to your prive e-mail, but it seems like your mail
service blacklists the whole of Hong Kong. :-\
Regards,
Steve
Michael Welter wrote:
Just when I thought I couldn't be wrong, I was wrong. We have
woodpeckers that drill into the arial telephone cables, and water
Critchfield wrote:
On Tue, 2004-04-06 at 10:49, Steve Underwood wrote:
Why do people get this uncontrollable urge to post, when the don't know
the correct answer? :-)
Having the absolute correct answer isn't always important if it steers
the requester in the right direction of self
Why do people get this uncontrollable urge to post, when the don't know
the correct answer? :-)
Regards,
Steve
Leo Ann Boon wrote:
I think that's the IMA ADCPM format.
Steve Underwood wrote:
Yves Chouinard wrote:
I found some posts regarding this issue dating of September 2003,
but no
real
Yves Chouinard wrote:
I found some posts regarding this issue dating of September 2003, but no
real answer.
The ADPCM format supported by Asterisk (the .vox files) is 4-bit, 8 kHz. I
need 4-bit, 6 kHz, which is also a widespread Dialogic format, to help
migration.
Is there an existing
Angus Berry wrote:
My business is somewhat trans-atlantic. Does anyone know if Asterisk's
DTMF processing is:
a) European OR US
b) European AND US
c) It's hardware dependent you fool.
d) here's a better answer
thanks :-)
The DTMF specs vary between countries, but not in ways that really
Alex Volkov wrote:
That would be great. Steve, please also consider using sourceforge.net to
host the project.
Alex.
OK, I considered it.
No.
Steve, first thanks for the great work (especially the bugfixes).
As development on SoftFAX/spandsp is especially fast and from the
source it appears
Hi Alex,
Alex Zarubin wrote:
Hi Steve and all,
1. Faxing from asterisk back to the same asterisk (from one Zap
channel to another)
doesn't work for us. Txfax called with the 'caller' parameter
issues CED, while the
receiving side needs CNG in order to switch to fax extension
Steve Underwood wrote:
Hi Alex,
Alex Zarubin wrote:
Hi Steve and all,
1. Faxing from asterisk back to the same asterisk (from one Zap
channel to another)
doesn't work for us. Txfax called with the 'caller' parameter
issues CED, while the
receiving side needs CNG in order
Reynaldo Simbulan wrote:
Hi Steve,
I am having this problem in which RxFax is still holding the file after
receiving a complete fax. Somehow the zap channel is still active but on the
fax client it was sent successfully.
If you call the line it is still busy.
There are disconnect issues if
Scott Laird wrote:
On Mar 31, 2004, at 12:01 AM, Nicolas Bougues wrote:
- hardware : you must have hotplug capable boards and bus. I believe
the only option there is CompactPCI, and of course cPCI boards and
chassis are very different from your day-to-day PCI stuff. Of course
Digium
Reynaldo Simbulan wrote:
Hi Steve,
I am having this problem in which RxFax is still holding the file after
receiving a complete fax. Somehow the zap channel is still active but on the
fax client it was sent successfully.
If you call the line it is still busy.
spandsp-0.0.1k.tar.gz and updated
Hi Alex,
Alex Zarubin wrote:
Hi Steve and all,
1. Faxing from asterisk back to the same asterisk (from one Zap
channel to another)
doesn't work for us. Txfax called with the 'caller' parameter
issues CED, while the
receiving side needs CNG in order to switch to fax extension
Hi Rob,
The modem training is not even close to correct. It looks like the FAX
software is not seeing good audio from the FAX machine. However, it
might be a strange problem in my modem software. The way to investigate
something like this is to look in spandsp-0.0.1/src/t30.c The first line
Hi all,
spandsp-0.0.1h.tar.gz seemed to get a lot more FAX machines talking, but
a number of people are getting rather high error rates on the images.
spandsp-0.0.1i.tar.gz addresses this, and should give much better bit
error rates for fax machines whose timing at the extremes the spec
Steve Underwood wrote:
Hi all,
spandsp-0.0.1h.tar.gz seemed to get a lot more FAX machines talking,
but a number of people are getting rather high error rates on the
images. spandsp-0.0.1i.tar.gz addresses this, and should give much
better bit error rates for fax machines whose timing
Hi James,
James Golovich wrote:
On Sun, 28 Mar 2004, Florian Overkamp wrote:
gcc -O2 -g -Iinclude -I../include -c -o app_rxfax.o app_rxfax.c
In file included from /usr/local/include/spandsp.h:40,
from app_rxfax.c:29:
/usr/local/include/spandsp/arctan2.h: In function
Eric Wieling wrote:
On Thu, 2004-03-25 at 09:33, Steve Underwood wrote:
exten = 5678,1,txfax(/tmp/testfax.tif|caller)
There are a zillion fax and tiff formats. I'm trying to figure out what
output format I should tell GhostScript to use. Any suggestions on
which format to try
Hi Reynaldo,
The is the only report of seg faults I have had with recent versions of
spandsp. There was an older version (spandsp-0.0.1b I think) which had a
silly bug that caused seg faults.
Some people have had older versions of libtiff installed, which seem to
cause seg faults. If you are
Hi all,
My SoftFAX is now up to spandsp-0.0.1h.tar.gz. This version has a number
of changes in the way the V.29 modem works. It also has some missing
functionality in the T.30 implementation filled in - it was not handling
EOM messages.
The previous version failed for several reasons with a
Alex Zarubin wrote:
Hi Steve,
We've got VFX/40ESC and VFX/40ESC plus, tried both. Can send faxes
from our
platform to your system if it helps.
After this problem is (hopefully) solved we are going to try FaxLab
http://www.qualitylogic.com/genoa_test_tools/fax/faxlab.html
Do you know of
Alex Zarubin wrote:
Hi Steve,
We've got VFX/40ESC and VFX/40ESC plus, tried both. Can send faxes
from our
platform to your system if it helps.
I have been testing with a VFX/40ESC. There is a chunk of the T.30 spec
I forgot to implement. I am adding it now. However, the biggest problem
with
Ah. That is a bug. audiofile.h should not be included by any of the
library source files. It should be fixed in the next update.
Regards,
Steve
Tilghman Lesher wrote:
On Monday 22 March 2004 18:26, Steve Underwood wrote:
Tilghman Lesher wrote:
On Sunday 21 March 2004 07:10, Steve
Hi Jon,
It seems spandsp sees nothing meaningful from the remote modem, although
the remote modem seems to see the initial message from spandsp. You are
the first to report this. I have no idea what might be wrong. If spandsp
received any kind of noise from the other end it would at least show
Hi all,
If you have had trouble with multiple concurrent channels running
app_rxfax or ap_txfax, where was a silly bug. Updated versions are
available at ftp://ftp.opencall.org/pub/spandsp
The latest spandsp-0.0.1f seems to working for quite a lot of people. I
guess there will still be plenty
Tilghman Lesher wrote:
On Sunday 21 March 2004 07:10, Steve Underwood wrote:
I have received more excellent problem report information, and I
have resolved a number of issues affecting my soft FAX machine when
working with various models of real FAX machine. The code now seems
to be working
, Line 1744
(ast_pbx_run): Extension 8663222818, priority 1 returned normally even
though call was hung up
Thank you.
Alex Zarubin
Webley Systems
-Original Message-
From: Steve Underwood [mailto:[EMAIL PROTECTED]
Sent: Monday, March 22, 2004 8:14 AM
To: [EMAIL PROTECTED]
Subject
Hi,
I have received more excellent problem report information, and I have
resolved a number of issues affecting my soft FAX machine when working
with various models of real FAX machine. The code now seems to be
working with a much greater range of fax machines. A problem affecting
the
Hi,
I have investigated some more fax machines that did not work with
spandsp, and made it more tolerant.
ftp://ftp.opencall.org/pub/spandsp/spandsp-0.0.1c.tar.gz is the result.
From what I have seen in today's investigations, I think this one will
work with considerably more quirky fax
Hi all,
It seems this week's release of spandsp fixed the major problems in the
previous release, but still people have had a lot of trouble. Working
with some of those who tried the software and gave me good feedback, I
have identified some apparently common bugs in fax machines, and I have
wangji wrote:
That's what Intel want to do, too. They guys have released a hardware
emulation software works like an four channel IP board. And they want to use
only interface board + host CPU instead of Dialogic products, so that they
needn't use DSP etc.
Have you actually tried playing
Michael Shuler wrote:
G.729 is just about everywhere. A lot of boxes use G.723 (and/or G.726) too
but G.729 ends up with about the same quality but at a much lower bit rate.
That's wrong. G.723 has the lowest bit rate amongst those codecs.
Most inexpensive hard phones don't use G.723 because
hank smith wrote:
hello I am just curious if there is any windows alternitives to Asterisk?
can I also use them with free world dialup?
thanks
hank
No, but maybe you could port Asterisk to Windows. No, that's not a joke.
The Zaptel drivers might be tough, but Asterisk's VoIP features would
Scott Stingel wrote:
If you'll be running commercial apps, I would recommend that you do a lot of
testing, especially load testing, with the types of applications you'll be
running. Dialogic boards, although incredibly expensive, do have lots of
horsepower built in for the purposes of encoding
If you want a freely usable implementation of SRTP look at
srtp.sourceforge.net.
Regards,
Steve
John Todd wrote:
I have found few VoIP clients that support encryption. The only one
that comes to mind is the Zultys devices (they have a softphone and a
hardphone that support SRTP.) I spoke
Robert Boardman wrote:
Hi
Just one question
do any of the Digium T1/E1 cards do DPNSS signaling?
Robb
Just one answer. No. :-(
Steve
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Jason Penton wrote:
Hi all
Does anyone know where I can get hold of the German 1TR6 ISDN signalling
protocol specification.
Thanks
Jason
Is that still used? I thought they were 100% CTR4 these days.
Regards,
Steve
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Asterisk-Users mailing list
Amaury Jacquot wrote:
Daniel Bichara wrote:
Hi Alex,
Alex G Robertson wrote:
Hi all,
I would like to have some information about your TE410p and TE405p
cards compatibility with telephony protocols adopted in Brazil.
- When in E1 mode, does it support R2 DIGITAL MFC 5C ?
You need a R2
WipeOut wrote:
This is an interesting statement in the press release..
SIPxchange, the industrys first open source based enterprise
communications suite, is grounded in the concept that a community of
ideas provides a more fertile ground for innovation, progress and
product development.
I
Dave Cotton wrote:
Just done a cvs checkout on 2 different machines 1 running Mandrake 9.2
with a 2.4.25 kernel the other Mandrake 10.0 and 2.6.3
the most 10.0 fails at this point
chan_zap.c: In function `handle_init_r2_event':
chan_zap.c:4773: error: too few arguments to function `zt_new'
Costa Tsaousis wrote:
Also I would turn off Hyperthreading (in the bios). It
may cause problems.
What problems? Are these digium H/W specific, asterisk specific or
generaly Linux problems?
I don't know if The HT problems are generic, or something quirky in the
Zaptel drivers. However, if
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
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
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. Yes, 60Hz is getting
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
Rich Adamson wrote:
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 probably have quite a lot of hum if it
300Hz is pretty high to filter out... it's still well within the rage
Jose Quinteiro wrote:
I live at sea level, and have never seen a woodpecker going at any
telco equipment, but have a 60Hz hum on my POTS line through my Adtran
750.
It goes away if I pick up the telephone I have cross-connected on the
same line. Could it be the same problem (i.e., tip-ring
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
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
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
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
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
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 :-) )
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
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 to the ITU specs.
Actually, nothing would use a 17Hz tone
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-modulated
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
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
the 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
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
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
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
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