asterisk users - Nov 2003 - DIAX 0.93 with some sound improvements and not only...

Hi all,

DIAX 0.9.3 is available for download from the same place:
http://www.laser.com/dante
or
http://www.geocities.com/tdanro

The new DLL contain the latest updates made by Steve in the iaxclient
library.
Still just IAX1 is supported (for the moment).

What's new in 0.9.3?

    - accept blank passwords;
    - accept for registration/calls host names, not only IP Address;
    - password no more displayed in clear in the registration window;
    - if no username is entered in the registration form, then 'guest'
is
used by default;
    - tooltip for memory buttons as full CallerID;
    - missed calls indicator display now the number of missed calls (in the
tooltip too);
    - tooltip for tray icon is now "You have X missed call(s)";
    - A bunch of signal processing stuff from speex (newly implemented in
the Steve K's iaxclient library, see
        http://iaxclient.sourceforge.net for more details):
        1) A denoising filter: This is a very effective filter which reduces
background noise;
        2) New AGC: This AGC implementation from speex seems to work much
better than the
            compander implementation used previously;
        3) A new "automatic" silence detection based on speex'
VAD. If you
set the silence threshold to
            a positive number, you now get speex' VAD, instead of my hacky
noise-threshold code;
        4) Bias removal: Removes bias from input signals. This makes other
DSP functions work correctly,
            including the VU-meter, which was often incorrect when bias was
present;
        5) Echo cancellation:. Speex echo is used (this cannot be changed
for the moment).

- Some bugs solved:
        - after clicking on VOL UP/DOWN only the left channel is heard;
        - tab stop in Registration form corrected


Please send me you feedback especially on sound quality (noise, agc, etc)
and bugs.

Best regards,
Dan

On Wed, 2003-11-12 at 15:07, Dan wrote:
> DIAX 0.9.3 is available for download from the same place:
Hi Dan :)

Do you know if anyone has successfully run DIAX on Linux with Wine?

After installing the VB6 runtime DLL, I ran diax.exe and got

fixme:ole:CoRegisterMessageFilter stub
fixme:ole:OLEPictureImpl_Construct Unsupported type 3
fixme:ole:OLEPictureImpl_SaveAsFile (0x404068d0)->(0x40406bc8, 0,
(nil)), hacked stub.
fixme:ole:VarParseNumFromStr (L"2",flags=80000000,....), partial stub!
fixme:ole:VarParseNumFromStr numparse: cDig=30, InFlags=954
fixme:ole:VarNumFromParseNum (..,dwVtBits=20,....), partial stub!
fixme:ole:VarParseNumFromStr (L"-99",flags=80000000,....), partial
stub!
fixme:ole:VarParseNumFromStr numparse: cDig=30, InFlags=954
fixme:ole:VarNumFromParseNum (..,dwVtBits=20,....), partial stub!
fixme:ole:OLEPictureImpl_FindConnectionPoint tried to find connection
point on {33ad4ed2-6699-11cf-b70c-00aa0060d393}?

and then a 'Runtime Error '6': Overflow' dialog with
'OK' ..

I don't know if any of these messages are even remotely useful, but I've
included them for completeness :)

Cheers,
Gavin.

---------- Original Message ----------------------------------
From: "Dan" <dtoma@fx.ro>
>Hi all,
>
>DIAX 0.9.3 is available for download from the same place:
>http://www.laser.com/dante
>or
>http://www.geocities.com/tdanro
Thank you for the update!  I have the following problems with it! When exiting
the program we get a General Protech error.  Also when calling Zap ports it
keeps ringing.  From DIAX to Sip it works fine!  It actually sound better then
before! But I can not call it from SIP get Audio missmatch.  I can call it from
normal Zap ports!

Hope this helps!  Keep up the work!

Hi,

----- Original Message ----- 
From: "Ariel Batista" <abatista@avionica.com>
To: <asterisk-users@lists.digium.com>
Sent: Wednesday, November 12, 2003 5:50 PM
Subject: Re: [Asterisk-Users] DIAX 0.93 with some sound improvements and not
only...


.....
> Thank you for the update!  I have the following problems with it! When
exiting the program we get a General Protech error.
This is a known bug (see the help file).... Hope to be solved when the IAX2
version will be available

> Also when calling Zap ports it keeps ringing.
Try to put a line in extensions.conf before the dial one

xxx,1,Answer
xxx,2,Dial(....
> It actually sound better then before!
The noise (the microphone one especially when used on a notebook) must be
drastically reduced now.
> But I can not call it from SIP get Audio missmatch.
What type of SIP phone?... I have test it with CIsco 7960 and it works as
expected..
Where did you gtet this message (on SIP phone or on DIAX)?

Best regards,
Dan