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
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
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
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
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 ste...@coppice.org
mailto:ste...@coppice.org wrote:
On 03/11/2014 12:36 AM
On 10/05/2013 11:07 PM, Darryl Moore wrote:
On 2013-10-04 5:36 PM, Steve Underwood ste...@coppice.org
mailto:ste...@coppice.org wrote:
On 10/05/2013 01:32 AM, Darryl Moore wrote:
I'll explain.
The g.729 compression algorithm is not protected by copyright, though
specific instances may
On 10/05/2013 01:32 AM, Darryl Moore wrote:
I'll explain.
The g.729 compression algorithm is not protected by copyright, though
specific instances may be. It is protected by a patent.
http://www.sipro.com/G-729.html
An open source version is available here:
http://asterisk.hosting.lv/
What
On 06/02/2013 11:02 PM, Chris Bagnall wrote:
On 2/6/13 2:01 pm, Muhammad Yousuf wrote:
I am trying to use asterisk as transcoder between voipswitch 2.0 and gsm
gateway. Voipswitch supports g723.1 but gsm gateway does not. Now I have
g723.1 codec in my asterisk. call leg from voipswitch is
On 03/15/2013 10:41 AM, Richard Kenner wrote:
There appears to be a disagreement between the encoding given in the
sources for Siren14 that are downloaded from Polycom (and the ITU, both
are the same) and that implemented by codec_siren14.so. The latter
agrees with the actual device.
If I make
On 10/09/2012 12:28 AM, Brett Lehrer wrote:
How many fax and voice calls (which codecs for tha latter ones ?) are on
average using your DSL line ?
1. Previously, I experienced failures during the process of converting
incoming PDF documents into ready-to-send fax image files while the reverse
On 10/07/2012 04:56 PM, Mikhail Lischuk wrote:
Gabriel Ortiz Lour писал 06.10.2012 17:07:
I am using this command to generate the TIFF from a PDF:
/usr/bin/gs -q -sDEVICE=tiffg3 -sPAPERSIZE=a4 -r204x196 -dNOPAUSE
-sOutputFile=$tiffFile -- $pdfFile
I use imagemagic's convert instead of gs,
it came from asterisk FAXing Howtos. Is
that correct?
I'll try doing some more tests with debug info ON and post back the
results.
Thanks,
Gabriel
2012/10/5 Steve Underwood ste...@coppice.org mailto:ste...@coppice.org
On 10/06/2012 02:53 AM, Gabriel Ortiz Lour wrote:
Hi
On 10/06/2012 02:53 AM, Gabriel Ortiz Lour wrote:
Hi,
Does anyone had the problem of asterisk SendFax + spandsp sending
only the first page of a multi-page TIFF file?
Seams to be related to spandsp ECM config.
Any thoughts about it?
Thanks,
Gabriel
Check the file with tiffinfo.
On 10/04/2012 09:29 PM, Brett Lehrer wrote:
I'm running Asterisk 1.8.11.1 and am connected to the nexVortex trunking
service over a DSL line solely dedicated to VoIP usage. For both incoming and
outgoing faxes, I'm getting a failure rate of just over 25%, and over a handful
of reasons.
Is
On 08/19/2012 11:45 AM, Lee Howard wrote:
On 08/17/2012 04:58 AM, Steve Underwood wrote:
On 08/17/2012 06:08 AM, Eric Wieling wrote:
Has anyone experimented with increasing the DAHDI chunk size in
improve fax reliability? If so, did it help, hurt, or not make any
difference?
I haven't
On 08/17/2012 06:08 AM, Eric Wieling wrote:
Has anyone experimented with increasing the DAHDI chunk size in improve fax
reliability? If so, did it help, hurt, or not make any difference?
I haven't found issues related to the DAHDI chunk size. The main thing
which used to hurt FAXing with
On 08/12/2012 10:32 AM, James Sharp wrote:
On 8/11/2012 8:05 AM, virendra bhati wrote:
Hi team,
I want to configure fax with asterisk. there a lot of fax link i found
by google but not working perfectly. my setup as follow
asterisk 10.x
centos 5.8
Want to used T.38 with SpanDSP...
Please
On 07/18/2012 09:43 PM, Matthew Jordan wrote:
- Original Message -
From: Alejandro Recarey a...@recarey.org
To: Asterisk Users Mailing List asterisk-users@lists.digium.com
Sent: Wednesday, July 18, 2012 6:30:26 AM
Subject: [asterisk-users] Using Asterisk 10.6 as a T38 Fax gateway
Hi
Hi David,
The old app_fax code, which allowed spandsp to be used with Asterisk
before Digium introduced the new modules supported the features you
want. Maybe someone can go through that code and port the feature into
the current res-fax code.
Steve
On 07/03/2012 09:57 AM, David Cunningham
extension (fax-rx, receive, 19) exited non-zero on 'DAHDI/i1/-4'
On Fri, Jun 22, 2012 at 12:25 PM, Steve Underwood ste...@coppice.org wrote:
On 06/22/2012 11:58 AM, Roi Stork wrote:
Hi,
Im able to send faxes with no errors, but the success rate for the
receiving side is less than 50%.
Asterisk
On 06/26/2012 10:24 AM, David Cunningham wrote:
Hello,
Does SendFAX have the ability to put the caller ID and timestamp on
the fax?
If so, is there a way to adjust the timezone used for the timestamp?
Thanks for any assistance.
SpanDSP has that ability, including per instance time zones,
On 06/22/2012 12:49 AM, Ahmed Munir wrote:
Hi,
I would like to know whether SpanDSP supports T.38 for Asterisk 10?
Because as far as using Fax for Asterisk, I'm getting some issues
using T.38
Only spandsp fully supports T.38 in Asterisk 10. The Digium module
cannot work in gateway mode.
On 06/22/2012 11:58 AM, Roi Stork wrote:
Hi,
Im able to send faxes with no errors, but the success rate for the
receiving side is less than 50%.
Asterisk usually returns records these errors as partial fax and fax
protocol error.
A lot of the error values returned by FAXOPT are 3RD_T2_TIMEOUT
On 05/17/2012 02:47 PM, gincantalupo wrote:
Hi Steve,
you are telling me there is no way to set a particular speed on my
iaxmodem in order to force the sender speed?
I have some problems with a customer who gets malformed faxes even if
no error occurs. Since I cannot tell the sender to lower
Hi Sebastian,
has still some issues that not all faxes pass ok, but does the work ==
still badly broken
Your log doesn't seem to show a spandsp error. It looks more like a bad
signal. Did you change anything else when you installed FFA? Usually
people move the other way to improve their
Hi,
On 05/16/2012 09:59 PM, Larry Moore wrote:
Read the subject line more closely.
Tested receiving too,
I set the Send Receive speed of the receiving analogue modem to that
below, the log file on the sending modem (iaxmodem) reported it
capable of 9600.
May 16 21:32:04.28: [ 2335]:
On 05/03/2012 10:35 PM, cjwstudios wrote:
If you're going full time hosted fax you will ultimately end up buying
a t.38/sip gateway like an Audiocodes Mediant.
Many people handling hundreds of thousands of FAXes per day would
disagree with that assessment.
On Thu, May 3, 2012 at 5:27 AM, Anita
On 04/20/2012 11:30 PM, Eduardo Pimenta wrote:
Hello all,
Does anyone know if EM over E1 signalling works on top of R2, ISDN
and where can I find a sample Dahdi configuration? Have done a lot of
google and cannot find a proper E1 configuration.
No it doesn't. EM signalling is the same
On 04/15/2012 07:26 PM, Patrick Lists wrote:
On 04/15/2012 01:15 PM, Gustavo Garcia Bernardo wrote:
Is it a good idea to use asterisk transcoding from G711 to iLBC or
should I find out any other solution not involving transcoding (f.e.
using G.729 that is supported in both sides). I'm worried
Hi Jean-Denis,
Your log shows the Mediatrix GW has problems. It sends a DCS signal to
the Asterisk box, but doesn't following it with TCF as it should. The
asterisk box times out waiting for TCF and tries to take recovery action
which fails.
Spandsp has some workarounds for bugs in
On 02/29/2012 02:28 PM, Dmitry Melekhov wrote:
btw, played with res_fax.conf
if I set maxrate=7200 fax machines try (and fail) 9600 anyway.
Why? If limited ti 7200? looks like bug...
Why do you think everything you don't understand is a bug? What you see
is correct behaviour. Any party in the
On 01/11/2012 02:39 PM, Olivier wrote:
Hi,
Maybe I missed it while checking it, but which spandsp version is
recommended to play with Asterisk 10 and T.38/T.30 gatewaying ?
I can see both spandsp-0.0.6pre17.tgz and spandsp-0.0.6pre18.tgz here
(http://www.soft-switch.org/downloads/spandsp/)
On 01/16/2012 03:59 PM, Roi Stork wrote:
Hi,
We noticed a very sharp drop in voice quality when using digium g729a
codec. The problem seems to happen if the A channel (caller's channel)
is a landline/mobile number contacted using the same outgoing provider
(as a local channel). It sounds like
On 01/13/2012 05:17 PM, mahesh katta wrote:
On Fri, Jan 13, 2012 at 1:58 PM, Ruben Rögels
ruben.roeg...@jumping-frog.org
mailto:ruben.roeg...@jumping-frog.org wrote:
Am 12.01.2012 18:50, schrieb mahesh katta:
I was search for free license but for this Digium require
purchase
On 01/11/2012 03:01 PM, Olivier wrote:
2012/1/5, Kevin P. Flemingkpflem...@digium.com:
On 01/04/2012 12:25 AM, Matt Darnell wrote:
Aloha,
We are looking to roll a solution that will have the following network
layout:
ISDN-PRI-- Asterisk-- T.38-- ATA-- Fax
Does version 1.8 with the
On 01/11/2012 11:16 PM, Olivier wrote:
2012/1/11, Steve Underwoodste...@coppice.org:
On 01/11/2012 03:01 PM, Olivier wrote:
2012/1/5, Kevin P. Flemingkpflem...@digium.com:
On 01/04/2012 12:25 AM, Matt Darnell wrote:
Aloha,
We are looking to roll a solution that will have the following
On 01/05/2012 07:45 PM, Michael Keuter wrote:
Am 05.01.2012 um 04:55 schrieb Matt Darnell:
On Wed, Jan 4, 2012 at 1:02 AM, David Klaverstyn
da...@klaverstyn.com.au wrote:
I'm using the Linksys PAP2T and the Grandstream with SpanDSP and tx_fax and
rx_fax on multiple installations with no
On 10/08/2011 02:50 AM, Kevin P. Fleming wrote:
On 10/07/2011 07:46 AM, Administrator TOOTAI wrote:
Hi,
I setup my first stock 1.8.7 asterisk (Ubuntu LTS 10.04 packages taken
from deb http://packages.asterisk.org/deb lucid main) including dahdi
from this same repository. No FFA involved.
On
On 10/08/2011 04:04 AM, Kevin P. Fleming wrote:
On 10/07/2011 02:20 PM, James Sharp wrote:
On 10/07/2011 12:27 AM, Nasir Iqbal wrote:
Check firewall and NAT settings!
Monitoring sip and media flow from asterisk cli can help, use sip set
debug on, rtp set debug on and udptl set debug on
No
On 10/09/2011 02:38 AM, Ryan Wagoner wrote:
On Sat, Oct 8, 2011 at 10:41 AM, Luke Hamburgl...@solvent-llc.com wrote:
Interesting. I just signed up with Gafachi (haven't even tested the service
yet) but I planned to make use of their T38 support since they are listed at
voip-info as being one
On 09/26/2011 01:01 AM, Bruce B wrote:
Paul,
These trolls are the people who put your kid to school and put food on
your table by giving valuable input and testing the open source software.
Are you sure Digium endorses this stand of yours? Does everyone at
Digium think the users who gives
Hi Tim,
On 09/01/2011 03:49 AM, Tim King wrote:
I realize that faxing is not great with voip but here is my confusion.
I have been working on a web based fax system for 2 weeks. During this
time I have sent over 100 2 page faxes without any errors. Now today
as things are finally completed I
On 09/01/2011 11:50 PM, Lee Howard wrote:
kirsten du toit wrote:
You should try disabling ecm..
This seems crazy to me. Why are you recommending it?
Because its the industry standard last resort of anyone who doesn't
understand FAX and is using T.38.
Steve
--
On 08/31/2011 01:15 AM, Fabian Borot wrote:
will installing spandsp help with t.38 pass-through?
The only part of spandsp which is relevant to T.38 passthrough is its
modem tone detection module, and I don't think the standard Asterisk
distribution can make use of that. Some people do use it,
On 08/01/2011 07:43 PM, Kevin P. Fleming wrote:
On 08/01/2011 04:12 AM, CB wrote:
On Thu, Jul 21, 2011 at 06:29:38AM +1200, CB wrote:
Are there any plans to include the ISAC codec in Asterisk? Is it
possible or
even desirable? Is ISAC open source (nothing indicates it is from the
WebRTC
On 06/21/2011 09:12 PM, khalid touati wrote:
Ok, for the variables, I can retrieve some of them like the caller
number and so on (I would assume that all the variables that last for
duration of call are there), but I still think that I sould not use
the h extension to continue after ReceiveFAX
On 06/20/2011 03:38 AM, khalid touati wrote:
Hi Guys,
I solved temporarely my issue by kind of tricking Asterisk, I used the
following line instead of the old:
exten = h,n,System('/usr/local/
bin/fax2mail -p -f ${FAXFILENOEXT} --cid-number ${CALLERID(num)}
--cid-name ${CALLERID(name)}
On 05/10/2011 12:55 AM, Olivier wrote:
2011/5/9 randulo rand...@randulo.com mailto:rand...@randulo.com
On Mon, May 9, 2011 at 2:20 PM, mgra...@mstvp.com
mailto:mgra...@mstvp.com wrote:
Lots of Android handsets support wifi, like my G2, aka HTC DesireZ.
Wouldn't ANY modern one
On 05/06/2011 02:09 AM, David Backeberg wrote:
T.38 has a boatload of problems, and most of those problems are
because people who aren't employed by Digium did not read the specs,
or they did read the specs, but felt like they had to violate the
specs to get their code to work with a different
Unless someone has broken something recently, you'll get better results
with spandsp than you get with the Digium FAX package.
Steve
On 05/04/2011 09:21 PM, Satish Patel wrote:
Did you try digim fax ?
Also you can record you incoming fax via mxmonitor and analize it.
--
Sent from my iPhone
On 05/05/2011 03:29 AM, Lee Howard wrote:
David Backeberg wrote:
On Wed, May 4, 2011 at 12:00 PM, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
(For my part, I'm actually surprised that nobody came up with a proper
protocol for encapsulating the stream of zeros and ones that make up
a fax
On 05/05/2011 01:07 AM, Tzafrir Cohen wrote:
Un-top-posting,
On Wed, May 04, 2011 at 10:01:37AM -0400, vip killa wrote:
On Wed, May 4, 2011 at 9:52 AM, Danny Nicholasda...@debsinc.com wrote:
*You are “Running before you learn to walk”! You can’t make T.38 work
(that’s ok, most other folks
On 04/21/2011 08:12 PM, Khaled W. Chehab wrote:
Dears,
I configured an account on my asterisk pbx to record the outgoing calls.
When the asterisk pbx user make a call and send a fax the call
recorded to wave file format.
I searched the internet and found a software that can play the
On 04/16/2011 08:47 PM, Ryan Wagoner wrote:
On Sat, Apr 16, 2011 at 1:56 AM, Steve Underwoodste...@coppice.org wrote:
On 04/16/2011 07:25 AM, Ryan Wagoner wrote:
On Fri, Apr 15, 2011 at 7:00 PM, sean darcyseandar...@gmail.comwrote:
Using spandsp-0.0.6-pre18, the Jan 22 release.
You
On 03/25/2011 04:58 AM, Thomas Winter wrote:
Hi list,
I have an 44100 Hz file with human voice, stereo with 16Bit.
When convertig this to 8 kHz, mono I loose a lot of quality and have
some ground noise. I tried several sox options but without success.
Can somebody help
best regards Thomas
Hi,
Has anyone seen G.711.0 in real world use? The spec was published quite
a while ago, but as far as I can tell there is no RFC defining the SDP
and RTP details needed to deploy it, and nobody advertises that they
support it in their products.
Steve
--
On 02/28/2011 10:12 AM, Stuart Longland wrote:
Hi all,
I've tried researching this, and so far, have struggled to find any
contemporary information on the issue, so I do apologise if asking this
irritates people who have answered this before.
I have managed to set up Asterisk 1.8 on the web
a plausible voice adjustment.
On Sat, Feb 5, 2011 at 9:44 PM, Steve Underwood ste...@coppice.org
mailto:ste...@coppice.org wrote:
On 02/06/2011 05:39 AM, Bruce B wrote:
Hello,
Are there any other other voice changer applications to
Asterisk other than the one from
On 02/06/2011 05:39 AM, Bruce B wrote:
Hello,
Are there any other other voice changer applications to Asterisk other
than the one from Lobstertech? (http://lobstertech.com/voice_changer.html)
Specifically interested in open-source but can have a look at
economical commercial alternatives as
On 01/22/2011 01:00 PM, Bryant Zimmerman wrote:
Where can I get the latest stable version of spandsp. That will work
with res_fax_spandsp.so. The link listed on the voip-info website is
dead. Any other location for download?
http://www.soft-switch.org/
There was a server failure. It should
On 01/21/2011 08:37 PM, Tom Rymes wrote:
On Jan 20, 2011, at 8:52 PM, Steve Underwood wrote:
A comparison wouldn't be complete without mentioning Hylafax. Hylafax has a
great infrastructure - tools for integrating with Windows clients, and so on.
Neither spandsp or the Digium FAX code can
On 01/20/2011 11:00 PM, Flavio Miranda wrote:
Hi all,
I realize that the application Receivefax can't handle with more than
one fax at the same time. In a environment with a lot of fax, some
caller get the signal but the operation can't be completed.
Is there a way to send busy tone to
On 01/20/2011 11:11 PM, Kevin P. Fleming wrote:
On 01/19/2011 02:30 PM, Bryant Zimmerman wrote:
On 01/19/2011 02:05 PM, Bryant Zimmerman wrote:
I am working on some fax tools for some of my users. I am reading the
https://wiki.asterisk.org docs for faxing.
Is see Application_SendFax and
On 01/21/2011 06:46 AM, Bryant Zimmerman wrote:
On 01/20/2011 11:47 AM, Steve Underwood
On 01/20/2011 11:11 PM, Kevin P. Fleming wrote:
On 01/19/2011 02:30 PM, Bryant Zimmerman wrote:
On 01/19/2011 02:05 PM, Bryant Zimmerman wrote:
I am working on some fax tools for some of my users. I am
On 01/17/2011 04:37 AM, Jeremy Kister wrote:
Since digium is apparently blind to users of their Free Fax for
Asterisk, does anyone have advice on how to report a crashing problem
with res_fax_digium and Asterisk 1.8.2 ?
Use spandsp.
I have detailed logs/reports and a backtrace ready, but I
On 01/08/2011 03:44 AM, Kevin P. Fleming wrote:
On 01/06/2011 11:34 AM, mgra...@mstvp.com wrote:
We should also be very clear that the Siren codecs are supported on the
Polycom SoundStation conference phones and the VVX-1500 Business Media
Phones. These codecs are not supported in the
On 01/06/2011 05:25 AM, Tim Panton wrote:
On 5 Jan 2011, at 13:07, Steve Underwood wrote:
G.722.1 is a 7kHz bandwidth codec. G.722.1C is a stretched version offering
14kHz bandwidth. These are most often found in Polycom phones, but they are
available elsewhere. The only widely supported HD
On 01/05/2011 03:29 PM, Bruce B wrote:
Hi Everyone,
1- Are the Siren7 and Siren14 the G.722 HD voice codecs?
2- Are these codecs only for Polycom units or are they universal
across all other SIP phones that advertise the HD voice codec like Aastra?
3- What is the main difference between the
On 01/06/2011 01:04 AM, Tilghman Lesher wrote:
On Wednesday 05 January 2011 07:07:10 Steve Underwood wrote:
On 01/05/2011 03:29 PM, Bruce B wrote:
Hi Everyone,
1- Are the Siren7 and Siren14 the G.722 HD voice codecs?
2- Are these codecs only for Polycom units or are they universal
across all
On 01/06/2011 12:05 AM, Kevin P. Fleming wrote:
On 01/05/2011 07:07 AM, Steve Underwood wrote:
On 01/05/2011 03:29 PM, Bruce B wrote:
Hi Everyone,
1- Are the Siren7 and Siren14 the G.722 HD voice codecs?
2- Are these codecs only for Polycom units or are they universal
across all other SIP
On 01/05/2011 02:39 AM, Tom Rymes wrote:
On 01/04/2011 8:55 AM, Kevin P. Fleming wrote:
On 01/03/2011 06:47 PM, Thomas Rymes wrote:
On Jan 3, 2011, at 3:22 PM, Kevin P. Fleming wrote:
On 01/03/2011 11:26 AM, Tom Rymes wrote:
[snip]
OK. Either way, though, the changes to echo cancellation
On 01/04/2011 09:53 PM, Kevin P. Fleming wrote:
On 01/03/2011 07:08 PM, Steve Underwood wrote:
On 01/04/2011 04:22 AM, Kevin P. Fleming wrote:
No. CNG tone is never used to affect the state of an echo canceller.
All G.168 compliant echo cancellers will respond to the CED tone
(generated
On 01/04/2011 04:22 AM, Kevin P. Fleming wrote:
On 01/03/2011 11:26 AM, Tom Rymes wrote:
Hi folks,
I was hoping that someone might be able to help clarify some confusion I
have on DAHDI Fax detection after spending some time searching. My
understanding is this:
I'll try.
1.) Echo
On 12/27/2010 08:05 PM, Elliot Murdock wrote:
Hello!
I am wondering how the differences between G729, G729a, and G729b
effect call bridging and server interoperability. For example, can
one server use the G729 code with another server that uses the G729A
codec?
Also, which version is Asterisk
Hi Michael,
Use spandsp. It is more relaxed about the file resolution, to avoid this
exact issue. Files with a resolution within 5% of 204x196 are accepted.
However, if you have really made the image width 1680 pixels, that is
wrong and I would be surprised if any FAX software accepts it.
On 10/09/2010 06:36 AM, Jeff LaCoursiere wrote:
On Fri, 8 Oct 2010, Bryant Zimmerman wrote:
I currently us Linksys/Ciscio, Grandstream and AudioCodes ata's. none
of the three perform well in all
enviroments. Between stablity issues, T38 and DTMF talkoff all three
suffer some
On 09/19/2010 12:06 AM, Darren Nickerson wrote:
On Sep 18, 2010, at 11:41 AM, Mark Deneen wrote:
On Fri, Sep 17, 2010 at 11:52 PM, Dean Collinsd...@cognation.net wrote:
Any thoughts on why the lack of traffic?
Cheers,
Dean
Not enough applications to play immature bathroom sounds.
You
On 09/14/2010 04:33 AM, Stanislav Korsei wrote:
Hello!
I've created clean installation of Asterisk 1.6.2.11 with spandsp 0.0.5.
When i try to receive fax I get:
Why install 0.0.5? Its ancient. the world has moved on.
[Sep 13 00:45:59] WARNING[3283]: app_fax.c:432 transmit_audio: channel
On 09/14/2010 04:23 AM, Joel Maslak wrote:
On Mon, Sep 13, 2010 at 1:12 PM, Hans Witvliet h...@a-domani.nl
mailto:h...@a-domani.nl wrote:
No these are also geo-stationary (same altitude, so same delay),
commercial and military satelites,
Yes, exactly. Geostationary satellites
On 09/07/2010 12:03 AM, Jeff Brower wrote:
Steve-
We are in the process of debugging a voice quality issue for a client of
ours that is a VoIP services provider. The client uses a softphone that
runs on a pjsip stack.
When placing a call using the softphone, it negotiates the use of G729
On 09/08/2010 03:23 AM, Gordon Henderson wrote:
On Tue, 7 Sep 2010, Tiago Geada wrote:
Hi,
I don't have any g729 codec license. But by reading Barry's complaint I get
to think that it is really unfair that Digium can't renew his license or
something.
I am a Debian user myself and I
On 09/06/2010 11:18 PM, Jeff Brower wrote:
Steve-
On 09/05/2010 04:08 AM, Vikram Ragukumar wrote:
Hello,
We are in the process of debugging a voice quality issue for a client of
ours that is a VoIP services provider. The client uses a softphone that
runs on a pjsip stack.
When
On 09/05/2010 04:08 AM, Vikram Ragukumar wrote:
Hello,
We are in the process of debugging a voice quality issue for a client of
ours that is a VoIP services provider. The client uses a softphone that
runs on a pjsip stack.
When placing a call using the softphone, it negotiates the use of
On 08/10/2010 09:40 PM, Jeremy Betts wrote:
I have always had very bad experiences with the x100p cards, they
always have very bad echo. If you need decent call quality I would
wait until you can afford a Digium card.
Use OSLEC with them, and they work OK. Even if they don't have a
On 08/10/2010 11:18 PM, Seann wrote:
Steve Underwood wrote:
On 08/10/2010 09:40 PM, Jeremy Betts wrote:
I have always had very bad experiences with the x100p cards, they
always have very bad echo. If you need decent call quality I would
wait until you can afford a Digium card.
Use
On 08/06/2010 04:43 PM, Jeff Brower wrote:
Steve-
On 08/06/2010 05:40 AM, Jeff Brower wrote:
Miguel-
El 05/08/10 14:50, Tim Nelson escribió:
- michel freihamich...@gmail.com wrote:
Dear Sir,
I tried to convert ilbc to ulaw and get the same result...Bad Voice
Quality
Regards
On 08/07/2010 03:15 AM, Jeff Brower wrote:
Steve-
El 05/08/10 14:50, Tim Nelson escribió:
- michel freihamich...@gmail.comwrote:
Dear Sir,
I tried to convert ilbc to ulaw and get the same result...Bad Voice
Quality
Regards
Again, iLBC is poor quality to begin with. You can't
On 08/06/2010 05:40 AM, Jeff Brower wrote:
Miguel-
El 05/08/10 14:50, Tim Nelson escribió:
- michel freihamich...@gmail.com wrote:
Dear Sir,
I tried to convert ilbc to ulaw and get the same result...Bad Voice
Quality
Regards
Again, iLBC is poor quality to begin with. You can't
On 07/26/2010 11:57 AM, Alexander Aksarin wrote:
On 20:59 Fri 23 Jul , Steve Underwood wrote:
That's just how your images look for me, so I guess your problem is
described here http://www.soft-switch.org/spandsp_faq/ar01s09.html
Steve
Big thanks for your help, Steve. I tried feh
On 07/26/2010 10:55 PM, Tzafrir Cohen wrote:
On Mon, Jul 26, 2010 at 09:54:24PM +0800, Steve Underwood wrote:
On 07/26/2010 11:57 AM, Alexander Aksarin wrote:
On 20:59 Fri 23 Jul , Steve Underwood wrote:
That's just how your images look for me, so I guess your problem is
described
On 07/23/2010 11:17 AM, Alexander Aksarin wrote:
On 21:46 Thu 22 Jul , Steve Underwood wrote:
It might help if you explained what you expect those pages should look
like. I see three quite plausible pages.
I expect to see this http://imagebin.ca/img/Eihpy0.jpg
That's just
On 07/22/2010 12:15 PM, Alexander Aksarin wrote:
On 09:06 Thu 22 Jul , Alexander Aksarin wrote:
Hello to all. I have succesfully received fax by app_fax, but tif files are
weird.
There a faxes sended by several fax machines to asterisk.
http://filebin.ca/hnnumf/122.tif
On 06/29/2010 05:35 PM, Gareth Blades wrote:
Zhang Shukun wrote:
hi, all
after a long time development, i need to deploy a production system.
i want to install latest Asterisk 1.6.2.9 on Centos 5.4 . one thing confused
me.
my computer hardware support 64 bit OS.
my
On 06/16/2010 11:31 PM, Karl Harris wrote:
On voip-info I found a few dated references to TDD support being in
the alpha stage and buggy.
Can anyone direct me to any newer information on this option?
There are installations where the TDD support in spandsp has been
integrated with
On 06/16/2010 11:44 PM, Danny Nicholas wrote:
I’m supposing that it is
1. no better or worse than SMS support
What relevance does SMS support have to TDD/TTY support?
1. dependent on the version you are on
I don't think the TDD support has been touched for years, so I doubt the
On 06/04/2010 02:27 AM, Kyle Kienapfel wrote:
http://en.wikipedia.org/wiki/G.729
Looks like theres A and B and no A/B so theres nothing to worry about
What's the point of quoting a page, if you are not actually going to
read it?
On Thu, Jun 3, 2010 at 9:09 AM, Alejandro Cabrera Obed
On 05/25/2010 07:54 PM, Kevin P. Fleming wrote:
On 05/25/2010 05:48 AM, Alexandru Oniciuc wrote:
Hello List,
I think I’ve discovered a little bug in t.38 bug in
1.6.0.22 regarding the speed (T38MaxBitRate) used to send the faxes.
Asterisk always
On 05/13/2010 10:48 PM, William Stillwell (Lists) wrote:
Ok, I ended up upgrading 2 of my 5 boxes to 1.6.2.7 , and using spandsp
0.0.6pre17, dahdi-linux-complete-2.3.0+2.3.0 , and enabled app_fax.
Hint: you need to install spandsp then run ./configure then make menuselect
:)
I was able to
On 05/12/2010 08:46 AM, David Backeberg wrote:
On Tue, May 11, 2010 at 3:30 PM, William Stillwell (Lists)
william.stillwell-li...@ablebody.net wrote:
Anybody know a reliable fax solution for 1.4.30 branch?
I am using PikaFax on another server and works very well (about 3000 faxes
a
On 05/08/2010 08:15 AM, Steve Totaro wrote:
On Fri, May 7, 2010 at 2:01 PM, Martin asteriskl...@callthem.info
mailto:asteriskl...@callthem.info wrote:
On Thu, May 6, 2010 at 3:11 PM, Steve Totaro
stot...@totarotechnologies.com
mailto:stot...@totarotechnologies.com wrote:
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