directly from the library
wiax.dll ( the one used by DIAX), which is a wrapper over the library and
can be used from VB too.
Best regards,
Dan
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.
Best regards,
Dan
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to be a bug in 0.9.10f. was great in 0.9.10e and is working
great in the 0.9.10g I have been testing for Dan. I'll push him a message
and see if he can update it to the web site.
James-
A newer version (0.9.11a) is under testing now. I hope to be able to post it
on my site
later today.
Best regards
to the registration form in French language
' - corrupted main form position if closing the app when minimized
' - exiting application during a call does not close that call first
' - cannot call using diax://number/alias when diax is already open
Pls send me your feedback.
Best regards,
Dan
/uLaw.
Best regards,
Dan
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Hi,
Solved, thanks
Dan, I'm just performing same tuning testing for echo.
Using Diax I can hear echo in my mobile. It happens for any codec that I
use (G711, GSM, iLbc, speex). Using Firefly with GSM or alaw, echo is very
reduced. (I can hear just a bit of echo). Can you suggest same adjustment
Hi,
disabling AGC echo is reduced.What AGC is and what is it for?
It is Automatic Gain Control for the local microphone.
It can be usefull when is a good isolation between speaker and
microphone and you want to capture low level sounds.
Best regards,
Dan
any sound in my ear phone.
It sounds totally dead.
Check that the mic is selected as record device and wave as playback device.
Check that wave is unselected as rec device and mic unselected as playback
device.
Best regards,
Dan
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Hi,
Dan,
Your suggestions solved my echo problems right away. The quality is very
good on both ends of my connection.
Thanks a million!!!
Fletcher Jones
Glad that I can help.
Keep on eye on this list.
Till the end of the year version 0.9.9f will be available for download, full
of new
.
- call volume - no counter incrementing
- audio configuration with different sound device for playback and ring
- Missing MSSTDFMT.DLL in WinXP SP2 and some Win98 systems
- no need to close the application in order to save the debug log file
Thank you all and a Happy New Year!
Dan
.
The source file for the wiax.dll is available for download too.
Best regards,
Dan
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send me your feedback.
Best regards,
Dan
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and the new VB runtime
libraries.
Please download the files again if you have one of those problems on your
system.
They are now 3 locations available:
http://www.laser.com/dante
http://www.geocities.com/tdanro
http://www.cosmica.ro/dante
Thank you again for your feedback and best regards,
Dan
a regular phone.
Thank you and best regards,
Dan
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,
Dan
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the app from the systray
- X10 send error if CM11/12 interface has some commands in the receiver
buffer
- error if trying to delete for the second time the log file
- unexpected crashes when registered with IAXTEL and/or other remote servers
As usual, please send me your feedback.
Best regards,
Dan
to solve this issue!?
Have you trioed to play with the 'Latency' parameter
in Audio Configuration form?
Try between 40 and 200.
IAXCOM I think use the default which is 200.
Best regards,
Dan
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info about your environment?
In the Control Panel Sound Configuration have you selected MIC as only input
for the used soundcard and wave out as the output?
Best regards,
Dan
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the echo test just press '#'. You can still leave me a message after
that.
Thank you ,
Dan
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is
much
more likely to produce the kind of results you describe than portaudio
latency
tuning.
DIAX 0.9.9g uses the latest available version of the iaxclient library.
They are other peoples on this list encountering the same audio lag
problem with DIAX 0.9.9g?
Thanks and best regards,
Dan
/diax/wiax_jb_old.zip
This use an older version of the jitter buffer.
Please send me your feedback.
Best regards,
Dan
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is the same independent of the codec used?
BR,
Dan
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Hi,
\ Em Sex 14 Jan 2005 16:43, Dan escreveu:
I dont have problems when calling PSTN extensions, and calling
VoceMail, EchoTest, etc. The problem is related with the conversation
between two DIAX Softphones.
Between 2 DIAX phone and the delay is in one direction only??
Yes. One direction only
, to be communicated with Asterisk PBX?
Have you tried DIAX?
It is a full featured IAX software phone, distributed as a freeware:
http://www.laser.com/dante
http://www.geocities.com/tdanro
There is an online help file too if you want to learn more about it.
Best regards,
Dan
Hi,
Is DIAX supported for G723 codec and can work on Windows OS?
It supports just: alaw, ulaw, gsm, ilbc and speex.
G723 is not very usual in the Asterisk world.
Best regards,
Dan
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http
/wiki-Codecs
to see a codec comparison.
You can try iLBC and Speex on DIAX.
Best regards,
Dan
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Hi Steve,
- Original Message -
From: Steve Kann [EMAIL PROTECTED]
On Jan 14, 2005, at 2:03 PM, Dan wrote:
Hi,
\ Em Sex 14 Jan 2005 16:43, Dan escreveu:
I dont have problems when calling PSTN extensions, and calling
VoceMail, EchoTest, etc. The problem is related with the
conversation
or even older:
http://www.laser.com/dante/diax/diax097a.zip
http://www.laser.com/dante/diax/diax096d.zip
http://www.laser.com/dante/diax/diax095.zip
and see if the problem persist.
If not, then it must be something in the new library and we will dig
further.
Thank you and best regards,
Dan
P.S. Pls tell
Hi,
From: Whisker, Peter [EMAIL PROTECTED]
GSM Codec is 13k plus overhead. That may work?
Peter
From: Bilal Ghayad [mailto:[EMAIL PROTECTED]
Well, what u advise us to use if the bandwidth is about 22kbps (dial up
connection in very old countries)?
Using iLBC you get the best results on low
/diax095.zip
and see if the problem persist.
Thank you and best regards,
Dan
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Asterisk
version and if you have or not this problem when calling between 2
DIAX clients.
Best regards,
Dan
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in any circumstances the following:
CVS-D2004.09.20.21.00.00-11/11/04-16:24:55
It can be related to this?
Best regards,
Dan
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. The same for the longer one.
How can I do to make it dial imediately when 3 digits starting with
1 are received?
Thank you and best regards,
Dan
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this behaviour here and as I'm
behind
a NAT I cannot use 2 DIAX phones connected to an external Asterisk
server (or there is a workaround for this?).
Thank you,
Dan
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Hi Florian,
- Original Message -
From: Florian Overkamp [EMAIL PROTECTED]
On Sat, 2005-01-22 at 11:46 +0200, Dan wrote:
I have a Cisco ATA186 connected to an Asterisk Server (SIP)
The dialplan uses 1XX for local extensions and XXX for
external numbers, where the first digit is always
to 9 seconds.
DialPlan:*St4-|#St4-|911|1#t8.r9t2-|0#t811.rat4-|^1t4#.-
Thank you.
Is working now.
Best regards,
Dan
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Hi Denis,
- Original Message -
From: Denis Galvão - iSolve [EMAIL PROTECTED]
Em Sáb 22 Jan 2005 07:51, Dan escreveu:
Hi all,
There is someone on this list having latency issues with DIAX who can
do this trace? I'm not able to dupplicate this behaviour here and as I'm
behind
a NAT I cannot
Hi All,
How can I detect DTMF tones inside Asterisk during a call?
What I want to get is the following:
- call console (OSS/dsp)
- use DTMF tones to adjust the MIC level for the OSS and
run scripts.
Any suggestions?
Best regards,
Dan
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Hi,
Hey I tried DIAX today and the speech quality was rather poor compared
to X-lite.
Which codec have you used?
The speech quality must be the same for the same codec.
Best regards,
Dan
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Hi Denis,
- Original Message -
From: Denis Galvão - iSolve [EMAIL PROTECTED]
Hey I tried DIAX today and the speech quality was rather poor compared
to X-lite.
Dan, do you know wich iaxclient version firefly is build on!?
I got better results(voice quality) using firefly, doesn't
to be used.
Thank you and best regards,
Dan
P.S.
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Hi,
http://www.cosmica.com/dante/diax/diax099i.zip
Sorry... the correct address is:
http://www.cosmica.ro/dante/diax/diax099i.zip
Best rregards,
Dan
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.
It is not a shortcut.
'#' uses the Standard asterisk transfer function and Transfer button an
internal implemented unattended IAX transfer function...
I'll take into consideration to make a configurable shortcut for the
Transfer
button.
Best regards,
Dan
to solve that delay issue soon.
P.S.: Someone forgot to say that DIAX supports USB Phones with /u flag too!
In the 0.9.10a version will support the Yealink USB phone too, including the
display, selectable ring tones, etc.
Best regards,
Dan
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not sure what i configured is right also? how to proceed
Thanks Regards
Pradhip
Try the examples from the DIAX help file.
BR,
Dan
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using '#' key (common way) or
using the Transfer Button.
Have you tried both of them?
Best regards,
Dan
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notification of the busy status (either in Mediax or Diax.
With an X100P card I get the PSTN line ringtone and/or busy tone in DIAX
when an outgoing call is in progress.
No need to have an Answer before...
BR,
Dan
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problems, drop me a mail.
Best regards,
Dan
http://www.laser.com/dante
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regarding the new and/or modified features, send me a
mail.
Thank you again for your help and best regards,
Dan
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send me your feedback.
Best regards,
Dan
P.S. Pls take into consideration that this version is not fully tested,
so use it with caution.
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of a
call.
Best regards,
Dan
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Hi Denis,
With this version I cant use my ATCom usb phone.
I didnt see it at the USB Phone options at the DIAX softphone menu. Only
yealink and eutectics.
I'll fix this ASAP and be back with an update.
Best regards,
Dan
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). No errors in the Asterisk console.
I have tried to search through the archive but ... nothing related to
this
There is any way to enable something like 'iax2 debug' but for Zaptel
channel?
Any suggestions are welcome.
Thank you and best regards,
Dan
Hi All,
As my previous mail was not posted on the list for more than 10 hours now,
I'll try to resend it.
Thank you,
Dan
- Original Message -
From: Dan [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, February
). No errors in the Asterisk console.
I have tried to search through the archive but ... nothing related to
this
There is any way to enable something like 'iax2 debug' but for Zaptel
channel?
Any suggestions are welcome.
Thank you and best regards,
Dan
this big and increasing delay?
Best regards,
Dan
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application CallMe feature.
Thank you and best regards,
Dan
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one and this is
all.
Please test it and send me your feedback.
I intend to release a new DIAX version this week. It will contain an updated
help file and some minor bug fixes.
Thank you all for your help.
Best regards,
Dan
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is disconnected after 20-30s?
Can you provide more details?
Thank you and best regards,
Dan
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.
As this is something for developers, let's move this thread outside the
list.
I'll send you a direct mail with the answer.
Best regards,
Dan
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. How can you
select it?
Some features are very cool:
- MWI
- IAX native Transfer
Keep up the good work.
Best regards,
Dan
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GSM. So the correct way to do it is:
disallow=all
allow=gsm
Best regards,
Dan
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you at the requested hours.
In this way I can easily help you solve all your technical problems
regarding DIAX.
Thank you for your understanding,
Dan
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Hi Andrew,
- Original Message -
From: Andrew Thompson [EMAIL PROTECTED]
Dan,
I was trying to find out if this patch was for the problem I described
several weeks ago:
No, this is only for the bug related with the following in IAX2: The phone
does not ring anymore after around 1min
be improoved in BT too.
The main issue with BT is that the standard is allmost the same for years.
The 'low power' feature of Bluetooth I don't think will be reached by
802.11 very soon.
Disclaimer: The value of predictions can go down as well as up :-)
Same for my comments..:-))
Best regards,
Dan
have setup multiple servers and try to dial a number which of the
servers is used to place that call?
I have defined just one (my box) and then tried to call an existing
extension, but it doesn't work.
Best regards,
Dan
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?
Using the BlueTooth connection to the phone you can:
- dial a number
- get the ring for the line
- answer a call
I'm sure that the Linux experts around here are able to build such a
channel.
This will be the cheaper solution to connect Asterisk to a GSM network.
Best regards,
Dan
.
The majority of the callers left interesting messages and even set
the callerID to something useful, like a PSTN/FWD/IAXTEL
phone number or even an e-mail address, but they are still a few
during the day with 'empty' calls.
Best regards,
Dan
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where the problem...
Have you tried DIAX with both IAX and IAX2 and it is the same issue?
BR,
Dan
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on one
of the phones network.
It works for me.
One of the NATs is a Wndows RRAS and the other one is a hardware broadband
router from Netgear.
I have just opened the 4569 UDP port on the firewall in both direction (for
input it is forwarded to the * box).
BR,
Dan
with the re-registration which occurs at 60s).
BR,
Dan
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to support my provider type of callerid signal, but does not work
for callwaiting callerid too).
Thanks,
Dan
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to
be related with the re-registration which occurs at 60s).
Maybe those two problems are related to each other, i.e. IAX2 tries to
bridge the call ... (I have no idea what I am talking about)
I think that the call is allways passed through the * server if IAX(2) is
used.
BR,
Dan
the whole conversation.
As I told before, it works in my environment, so I have no way to reproduce
this behaviour here.
BR,
Dan
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and send me the log?
BR,
Dan
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this was something known.
They are some known problems with both IAX(1) and IAX(2) client libraries,
but nobody work on IAX(1) anymore.
BR,
Dan
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the need to
reregister?
Thank you and best regards,
Dan
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it.
Thanks a lot.
Best regards,
Dan
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'1234' is the extension allocated for DIAX
'name' is the registration username used for DIAX registration.
Best regards,
Dan
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for transfer and
ring).
Try the following model for user definition in iax.conf:
[mike]
type=friend
username=mike
secret=passwd
auth=md5
host=dynamic
callerid=Mike 407
context=local
It seems that you do not have the 'username' line inside definition.
Best regards,
Dan
else had this same problem.
You must change the dial command to something like that:
exten = 111,1,Dial(IAX2/user,20)
if you keep IAX/user then IAX(1) is used.
Check too if chan_iax2 is loaded.
BR,
Dan
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[EMAIL
Hi,
- Original Message -
From: Erick Weber V. [EMAIL PROTECTED]
Someone know wich is the best firmware for the ATA 186 with *
Version 2.16 (SIPH.323) works great for me (use it in production for more
than 6 months now without any problem).
BR,
Dan
Hi,
- Original Message -
From: Doug Harris
Where can I download this version ?
Cant find it here http://www.grandstream.com/TEMP/FIRMWARE/
Do you mean ATA-18x (from Cisco) or ATA-286 from Grandstream???
BR,
Dan
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further.
Best regards,
Dan
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to the audio path...
BR,
Dan
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, iaxComm, IAX Phone).
I am interested too, in order to know if something is broken after the
changes in the library.
BR,
Dan
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Hi,
- Original Message -
From: Matt Riddell [EMAIL PROTECTED]
Strange, I have no problem with echo!
:-)
This is normal, as it does. NOTHING..
:-))
It is just a checkbox in the interface (for the moment)...
BR,
Dan
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Hi,
- Original Message -
From: Matt Riddell [EMAIL PROTECTED]
also...
I cannot turn of the beep!
If I turn it off and go back, it's back on again!
You're right this is a bug.
Solved now
Thanks and best regards,
Dan
.
I'l try to solve them one by one, without disturbing the whole list.
Thank you,
Dan
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Hi,
- Original Message -
From: Matt Riddell [EMAIL PROTECTED]
Please ignore _ALL_ messages from me regarding problems in the last couple
of days.
I cannot do it because some of them are true DIAX bugs...:-))
Good luck,
Dan
___
Asterisk
If you have the package available for download for free from SJLabs, then
you only have G.711 codec installed on SJPhone.
If you are a developer, you can register for a G.729 codec from SJLabs.
BR,
Dan
P.S. Have you tried X-Lite? It has G.711ulaw, G.711a law, GSM and iLbc.
- Original
thing overall, but I still need something much cheaper for home
use.
Thanks,
Dan
- Original Message -
From: Markku Korpi [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, May 29, 2003 4:50 PM
Subject: RE: [Asterisk-Users] External FXO device (USB or ethernet),
supported by Asterisk
of the phone t the PSTN line.
Have you considered a S100U and one of those $35 FXS to FXO converters?
There is something like that? Where I can find such a converter and how this
thing works?
BR,
Dan
- Original Message -
From: Jim Flagg [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday
Hi,
Check to have a common set of codecs.
If X-Lite is used and at the other end is a phone
without GSM support, then it doesn't work.
Try to disable GSM on the soft phone (if
X-Lite).
BR,
Dan
- Original Message -
From:
Dave Alan Caruana
To: [EMAIL PROTECTED
.
Overall. this is something that make sense to use for some specific
purposes.
Thanks for this info,
Dan
-
- Original Message -
From: Jim Flagg [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, May 30, 2003 12:24 AM
Subject: Re: [Asterisk-Users] External FXO device (USB or ethernet
One more thing which can be a big issue with this device.
It hangs the line ONLY based on busy tone... if not correctly detected, then
it will keep the line open for ever, or you can select a call limit
(15/30min.)/
Dan
- Original Message -
From: Jim Flagg [EMAIL PROTECTED]
To: [EMAIL
Hi,
How can the dial tones on a CISCO 7960 be modified? Compared to the ATA
186,
I
could not find any settings that make a change possible.
Go to Settings SIP configuration 9 (Out of Band DTMF)
You can choose between avt, avt_allways and none
BR,
Dan
- Original Message -
From
Hi Dave,
If you have registered theSIP phone with
Asterisk, then you must have a line like:
exten = 555,1,dial(SIP/[EMAIL PROTECTED],52,153.207)
in extensions.conf file
Then call 555 from the SIP phone to access the
destination.
BR,
Dan
- Original Message -
From:
Dave
?
How can I do to pass all the calls through Asterisk, even if a codec
conversion is required or not?
Thanks,
Dan
- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, May 31, 2003 5:27 PM
Subject: Re: [Asterisk-Users] Passing audio stream
Many thanks,
Dan
- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, May 31, 2003 7:29 PM
Subject: Re: [Asterisk-Users] Passing audio stream through Asterisk or not?
On Sat, 2003-05-31 at 10:51, Dan wrote:
Hi,
if you turn off
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