Jan Rychter wrote:
Does G.729 provide better voice quality than GSM?
(a question for people who have tried both)
It depends. The bit rate of G.729 is a lot lower, so it starts with a
disadvantage. To overcome that, they made it a lot more complex and
tuned to the human voice. The result is for
Matthew John Darnell wrote:
Why hasn't someone found 50 people who sound alike, put them in sound
studios and record the 10,000 most commonly used words. You would all
differnent forms of the 1,000 most words, i.e. leading, trailing, question
etc.
You can synthesize the other 0.05% when you run i
Jeff Noxon wrote:
Many of you are familiar with how lousy Festival sounds.
AT&T has a product, NaturalVoices, that sounds much better. There are
male & female voice fonts for US/UK/Indian English, French, Spanish,
and German.
I am considering offering a linux-based text-to-speech engine based on
Moshe Yudkowsky wrote:
At 10:11 2003-07-16 -0700, Chris Albertson wrote:
if you want a synthetic voice to sound
natural you will have to tell the software the _intent_ of the words
not just the words. You would need a markup language for that
I said yes
The W3C has a TTS markup language,
LQ (Asterisk) wrote:
Dear fellows,
I need to use Asterisk with an E1 card with CAS R2 signalling for Argentina.
I know that the E100P don't support it right now.
Correct
Does anybody developing R2 drivers?
Yes.
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John Todd wrote:
LQ (Asterisk) wrote:
Dear fellows,
I need to use Asterisk with an E1 card with CAS R2 signalling for
Argentina.
I know that the E100P don't support it right now.
Correct
Does anybody developing R2 drivers?
Yes.
Interestingly terse reply; perhaps you can be more specific?
Hi Joe,
Most auto-dialers will accept commas in the dial string, and insert
delays where they occur. Will that work for you? Its normally used to
insert a delay after a 9 on a PBX, to get a stable outside line before
further dialing.
Regards,
Steve
Joe Antkowiak wrote:
Hi,
I am using a chann
That looks a bit like this one:
http://www.planet.com.tw/product/product_intro.php?menu_id=3
I've also seen it with Micronet and a couple of other well known
Taiwanese networking company's names on it. I don't know who makes it,
though.
It seems well made. Its pretty ugly (some would say "styl
David Boreham wrote:
P.S. Please do not answer again that this setup cannot work. In this
moment
I cannot accept such an answer.
Your e-mail made me chuckle. When I worked at Octel/Lucent
in the mid-90's we were constantly sniped at for trying to make
a voicemail system which ran on g
Scott Lambert wrote:
On Sat, Jul 19, 2003 at 01:01:52AM +0800, Steve Underwood wrote:
That looks a bit like this one:
http://www.planet.com.tw/product/product_intro.php?menu_id=3
rather expensive to me. These things have less DSP and compute to do
than an ADSL modem, and should cost no more
Hi,
I put a TE410P card in a machine (a Tyan 2665 with 2x2.4GHz Xeons). A
red flashing light circles around the 4 RJ48C sockets. I load the
wct4xxp driver, and the flashing light stops. Whether I connect an E1
signal or not, no lights are shown, and no alarms are reports in the
/proc/zaptel/XX
OK Funny guy,
Mark Spencer wrote:
I put a TE410P card in a machine (a Tyan 2665 with 2x2.4GHz Xeons). A
red flashing light circles around the 4 RJ48C sockets. I load the
wct4xxp driver, and the flashing light stops. Whether I connect an E1
signal or not, no lights are shown, and no alarms are rep
Q: What's the difference between Asterisk and a softswitch
A: About $100,000
Soft switch - Hard to afford!
Regards,
Steve
Bruce Ferrell wrote:
I've been working in the VoIP industry for just a bit over a year
now... Mostly taking care of the underlying systems. I've now reached
the point whe
Hi Matthew,
That argument doesn't seem to work. I don't hear many complaints here
about the cost of the VoiceAge codec. It's the clunkiness of the
protection scheme people don't like. It's only the protection scheme
that seems to be making people want to dump the VoiceAge code.
Remember how Mi
Eric Wieling wrote:
On Tue, 2003-08-12 at 15:37, Mark Spencer wrote:
Couldn't agree more. The G.729 codec is so unDigium-like... don't buy
it is my recommendation.
I don't think anybody buys G.729 just to have it. They buy it because
they *have* to have it. And we sell it because they
Scott Stingel wrote:
Hi all-
This question is for those familiar with EuroISDN setup.
I have a customer in Europe where I'm going to install an asterisk based
system with 4 E1's. The customer will configure them all in one large hunt
group.
My question is about the E1 channel configuration. I
John Todd wrote:
Are they selling the Grandstream? They don't seem to actually say
that; they only say that the Grandstream is the only phone they tested
(other than their own system? not clear.)
A picture paints a thousand words, they say:
http://www.sipphone.com/tiki-index.php?page=Order%20
Kim C. Callis wrote:
I was reading on www.vovida.org/applications/downloads/G729A/ (home of
VOCAL) pages, and that there is a free license use for non-commercial
for G.729A. Is that usable under Asterisk or strictly a Vovida offering?
This was a publicity stunt by VoiceAge, which Cisco/Vovida seem
Steve Underwood wrote:
The ITU G.729 code is pretty much useless for real world use. It is
very slow. It gets the right answers, but not by efficient means. All
the voice codec reference code I have seen is like this. The people
who develop these things *have* to write an efficient version, as
Dan wrote:
Hi Steve
Steve Underwood wrote:
06.10 isn't that great a codec,
though. I don't think it is used very much on the GSM networks these
days. Most of the time they use the enhanced full rate (EFR) or half
rate codecs.
What do you mean by "isn't a great cod
Kim C. Callis wrote:
I was trying to do a little searching to see if there has even been a
comparison between Asterisk and VOCAL or any of the other OSS packages?
"Practical Voice Over IP using VOCAL" published by O'Reilly and
Associates, attempts to make a strong case about how scalable VOCAL. Of
Hi Dan,
Dan wrote:
- Original Message -
From: "Steve Underwood" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, August 13, 2003 9:49 AM
Subject: Re: [Asterisk-Users] Open G.729A codec
Steve Underwood wrote:
After writing this I got curious about
Brian West wrote:
bkw
PS: The worse that can happen is it can't be used in court.
I though that tended to be the best that could happen :-)
Regards,
Steve
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Paulo Mannheimer wrote:
Hi,
I'm testing an E1 with E&M 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 configure
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 ear
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.
Rega
, 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 Under
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 c
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 tr
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 Ya
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 fre
ilto:[EMAIL PROTECTED] On Behalf Of LQ
(Asterisk)
Sent: September 11, 2003 10:19 AM
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Is there any MFC-R2 implementation for
asterisk?
The last thing that I read about it was:
Steve Underwood [EMAIL PROTECTED] wrote on Sep 3:
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,
Steve
Hi Brad,
If you want to detect that a sound is voice, rather than something else,
it isn't easy. There is information around on the Internet about
methods, but I have never tried them and don't know how well they work.
Unless you have some understanding of DSP I wouldn't bother trying. On
the
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:
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 passthroug
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 th
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
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
___
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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 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 softwa
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
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 con
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
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 i
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 Hi
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
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
t; This should be all that is necessary to run a T.38 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 (so
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 grou
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 re-enco
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 acti
of app_fax
>> to work with Asterisk 1.4x ? Someone has tried it ?
>>
>> Best Regards,
>> Fernando
>>
>> Thomas Kenyon wrote:
>>
>>> Steve Underwood wrote:
>>>
>>>
>>>>>
>>>>>
>
On 03/11/2014 12:36 AM, Mike Diehl wrote:
Hi all,
For the most part, we are finding that Fax for Asterisk works pretty
well. However, we have seen some wierdness that we'd like to try to
fix.
Once in a while, we will get a partial result report for a given fax
but when we look at the actual .t
that
appears to only be a single page.
But, since FFA isn't providing acknowledgement, the sending fax
machine is resending the document multiple times!
Mike.
On Mon, Mar 10, 2014 at 12:49 PM, Steve Underwood <mailto:ste...@coppice.org>> wrote:
On 03/11/2014 12:36 AM, Mi
Hi Jeff,
On 04/08/2014 12:13 PM, Jeff Brower wrote:
Darrel- The G729 essential patents were *granted* in 1996, but applied
for prior to June 8 1995. That means their lifespan is either 20 years
from their application date, or 17 years from their grant date,
whichever is greater (http://www.usp
On 04/09/2014 06:54 PM, Tzafrir Cohen wrote:
On Wed, Apr 09, 2014 at 10:19:59AM +0800, Steve Underwood wrote:
Hi Jeff,
On 04/08/2014 12:13 PM, Jeff Brower wrote:
Darrel- The G729 essential patents were *granted* in 1996, but
applied for prior to June 8 1995. That means their lifespan is
On 03/30/2016 08:23 PM, Vitor Mazuco wrote:
Hi!
Is possible to use X100p TDM400P, Tdm410p, Tdm400, A400p, Ax400p or
any others digium card FXO for use Fax modem?
Thanks.
Asterisk + iaxmodem gives you a bunch of soft FAX modems. Add one of the
analogue PSTN interface cards you listed and you h
Alastair Maw wrote:
Fax uses FSK modulation to transmit the data. If you compress this in
a lossy way (GSM, MP3, whatever) then the integrity of the data is
affected (more or less seriously depending on the codec used). Fax
machines are generally quite picky, so compressing faxes is unlikely
t
ProvoCityPower wrote:
The question asked here, "why on earth you want to push fax data over
a VoIP link at
all. Fax compression isn't very efficient." may speak volumes about
the future role of VOIP. My plans are to role out a VOIP connection to
thousands of Customers. Many have legacy fax equi
Walker Haddock wrote:
On Mon, Dec 15, 2003 at 05:06:57PM +0200, Dan wrote:
Hi,
- Original Message -
From: "Walker Haddock" <[EMAIL PROTECTED]>
Dan, you say fax works better on the TDM400 than the ATA186. I'm having
problems with faxing on the TDM400.
I had to drop t
Tilghman Lesher wrote:
On Monday 22 December 2003 10:12, Philipp von Klitzing wrote:
Hi!
I'm also curious if anyone else is doing this or if anyone else
is using the Asterisk TDD support.
Excuse my ignorance: What exactly is TDD? Is it US specific?
It's a specification for sen
Steve Totaro wrote:
Supports H323
http://www.viseon.com/prod/c_VisiFone.asp?id=133
So? Whilst there are still only a few VoIP audio phones available,
almost every computer related manufacturer in Asia has at least one
video phone model like this. There must be dozens of units like this
availa
Eduardo Goncalves wrote:
On Mon, 22 Dec 2003 15:48:37 -0600
Steven Critchfield <[EMAIL PROTECTED]> wrote:
asterix:~# modprobe tor2
Zapata Telephony Interface Registered on major 196
Detected Tormenta 2 Quad T1/PRI or E1/PRA at 0xfe121800/0xfe121000
irq 7 Did not get DONE signal. Short file mayb
Steven Critchfield wrote:
On Wed, 2003-12-24 at 19:11, Mahoney, Matt wrote:
Hi,
Does asterisk support any kind of voice encryption?
Not right now. As I understand it, it is a problem with the fact that
each packet would have to be able to be decrypted even if packets in the
stream are los
And SRTP uses AES!
Steve
Brian West wrote:
I understand AES can do this.
bkw
On Thu, 25 Dec 2003, Steve Underwood wrote:
Steven Critchfield wrote:
On Wed, 2003-12-24 at 19:11, Mahoney, Matt wrote:
Hi,
Does asterisk support any kind of voice encryption?
Not
That status page tells you the porject has gone nowhere so far. What you
need is not a driver. It is a development project!
Regards,
Steve
Ray Burkholder wrote:
Current Status: http://www.openss7.org/asterix.html
Ray
Do I need a special Digium Card (E100-SS7) or use my E100P card and compile
Patrick Wong wrote:
Hi all,
I just checked out that Asterisk which is a platform I am interested
of. I would like to install it to the Linux box for a trial. I have
some legacy Dialogic hardware on hand, don't know they will work with
Asterisk or not. For analog loop start interface I have
WipeOut wrote:
Granted five 9's is never easy but in a cluster of 10+ servers the
system should survive just about anything short of an act of God..
You do realise that is a real dumb statement, don't you? :-)
A cluster of 10 machines, each on a different site. Guarantees from the
power company
9's allows about 6
minutes downtime a year. That means 100% of all failures must have
automated failover, as manuals repair could never be achieved so fast.
Physical diversity if essential for that.
Regards,
Steve
Chris Albertson wrote:
--- Steve Underwood <[EMAIL PROTECTED]> wrote:
Anton Tinchev wrote:
Just spended ~ hour googling - all boards are based on GC-XX or I750X
Chipsets - all for Xeons. There also some boards for Pentium 3.
Can someone point me to a board with 64Bit 3.3v PCI for ordinary P4
with 800Mhz FSB.
Thanks
Unless one has appeared in the last couple of wee
Hi Daniel,
You will find libr2 is only about 10% of an implementation, and a bad
one at that. I now have 95% of a good implementation, but its not yet
released.
Regards,
Steve
Daniel Bichara wrote:
Hi all,
I will start testing libr2 for brazilian R2. Any clue?
Daniel
__
John Todd wrote:
hurricane
tornado
You missed typhoon!
Regards,
Steve
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lling differences between countries ( I am
trying to run .ar at .br)?
Daniel
Steve Underwood wrote:
Hi Daniel,
You will find libr2 is only about 10% of an implementation, and a bad
one at that. I now have 95% of a good implementation, but its not yet
released.
Regards,
Steve
Daniel Bichara
Anton Tinchev wrote:
Richard Grinnell wrote:
Dell - PowerEdge 400SC Server Under $300 with MIR
Intel® P®4 Processor at 2.8GHz, 512KB Cache, 800MHz
FSB
For those of you who aren`t familiar with the 400SC, this server is
an Intel i875P chipset based server
with an 8x AGP slot. It is compatible wit
Anton Tinchev wrote:
Steve Underwood wrote:
Anton Tinchev wrote:
Richard Grinnell wrote:
Dell - PowerEdge 400SC Server Under $300 with MIR
Intel® P®4 Processor at 2.8GHz, 512KB Cache, 800MHz
FSB
For those of you who aren`t familiar with the 400SC, this server is
an Intel i875P chipset based
John Brown (CV) wrote:
It appears that zttool doesn't actually report T1 span
errors.
If I inject BPV's, crc errors, framing errors, etc into
a T1 span, the counters on zttool don't change.
It works OK for me with Tormenta 2 and TE410P boards. Both zttool and
the /proc/zaptel/x files seem to
ublic carrier, and therefore have no idea where to start.
Would someone (Steve Underwood ;-) )mind at least putting me on the right
track so I can address this issue?
Thanks in advance Steve
Jason
I don't know why this would fail. An ISDN card should be properly
synchrinised to the PSTN, and use
TC wrote:
To complete this rather lengthy topic... what happens if you ignore all of
this and just slap a bunch of systems together with no regard to a master
sync source? The quality and stability of your network will likely not be
as good as what it could be. If your clocks (in each device) hap
Stephen Davies wrote:
On Wed, 14 Jan 2004, TC wrote:
What are the practical effects with in-correct clock sync
-like to you hear odd buzzing, or dropped voice or gaps of audio ??
Old-fart anecdote about this - in the early 80s we had some 1200bps
modems that we used to connect to client s
Rich Adamson wrote:
To complete this rather lengthy topic... what happens if you ignore all of
this and just slap a bunch of systems together with no regard to a master
sync source? The quality and stability of your network will likely not be
as good as what it could be. If your clocks (in each d
Rich Adamson wrote:
What are the practical effects with in-correct clock sync
-like to you hear odd buzzing, or dropped voice or gaps of audio ??
You may get gaps where frames are discarded, this will be across all timeslots
so an individual loss isnt a lot of data, you'll probably get awa
Hi,
My guess would be the lengths in the header are not set right. If a wave
file (or a file with a similar structure, like TIFF) works with some
things and not with others, the problem is usually the lengths in the
header. Some software just complains when the lengths are wrong. Some
tries to
Olle E. Johansson wrote:
LQ (Asterisk) wrote:
Hi guys,
I was reading that Steve Underwood is working on Asterisk R2 signalling
support, and has the 95% of the work done.
What is R2? I'm curious.
Half of R2D2, of course.
Its also a stupid clunky multi-tone based telephone signaling s
Eric Wieling wrote:
How does Grandstream become patent indemnified for their hardware? I
would assume they did not pay for a license for G723,1 and G729 directly
to the patent holding company. Maybe they did. I always assumed the
indemnification came with a DSP that implemented the codec.
Mo
Hi David,
David Liu wrote:
Hi there,
Anyone had any success deploying Asterisk with a T100P or T400P card
in Hong Kong? To my understanding, Hong Kong carriers only provide
IDA-P or IDA-M lines. I am looking to use IDA-P. Is this possible
with the card?
I know Cisco 2651XMV with a VIC ca
Martin Pycko wrote:
You have to contact www.openss7.org. The site may look dead but they
sell ss7 together with asterisk.
Yes and no. The sell access to the SS7 CVS. It does not work with
Asterisk. There is a project page about OpenSS7 - Asterisk integration,
but it is a project that never we
Jan Czmok wrote:
Michael Devenijn ([EMAIL PROTECTED]) wrote:
Jan,
Where can we get any technical documentation about sccp protocol i've searched with google and at cisco but i don't find anything useful ...
The only useful resource is imagination :-)
Skinny is a Protocol developed by Se
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
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
sounding.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Steve Underwood
Sent: Sunday, 25 January 2004 4:17 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk Indications
The correct tone is 400*17 (383 + 417) according
Christopher Lee wrote:
On Sun, 25 Jan 2004, Steve Underwood wrote:
Actually, nothing would use a 17Hz tone - it doesn't pass through a
300-3400Hz channel very well :-)
It's not a 17Hz tone. Australian (and others) tones are single-frequency
tones that are amplitude-modu
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
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
TC wrote:
Mythical Asterisk Creatures, oft-discussed, rarely seen:
1) An "advanced" graphical user interface
2) An IAX2 hardware device
3) A Radius CDR report module
4) A live-method, robust SQL-based dialplan
5) LDAP/SQL/Radius authentication for SIP phones
6) Robust R2 signalling suppor
Klaus-Peter Junghanns wrote:
Hi Martin,
libpri misses all the fun stuff :-(
hold, retrieve, suspend, ect, cd, conf, 3pty ..
but i am going to change that :-)
regards
kapejod
It misses all the timers, too. :-)
Regards,
Steve
___
Asterisk-Users m
Hi all,
I would like to announce the availability of an initial test version of
a totally software FAX facility, suitable for use with Asterisk. This is
a first public test release, so don't expect a solid polished product
just yet. People have shown interest in what I am doing, and here is the
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
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