yes please install our ssh key and send me the details.

the key is in the svn tree in

ssh_keys/shinzon.pub

you can find us on irc all day or im or email
me.

 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale@hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org
pstn:213-799-1400


----- Original Message ----
From: "freeswitch@dalethatcher.com" <freeswitch@dalethatcher.com>
To: freeswitch-users@lists.freeswitch.org
Sent: Monday, January 28, 2008 8:01:00 AM
Subject: Re: [Freeswitch-users] Audio rate changes and clipping on playback

I mean it sounds like parts of the sound are missing.  e.g. the
frequency of playback seems to change and I can hear a warbling/slow
down from the Freeswitch instance on the audio samples.

If you (or any others on the list) have a spare couple of minutes then
I'm happy to send you the details of my test server.  It would be
reassuring if someone else could hear and verify the issue.

thanks,

- Dale

On Mon, Jan 28, 2008 at 02:05:20PM +0100, Sluschny, Thomas wrote:
> don't know what you mean with clipping, but I've notice clipping errors when I play sound files with maximum levels with G.711, clipping means that on max. posivite value a max. negative value is played.
> this is true only on G.711, e. g. Linear16 codes are fine.
>
> regards,
> thomas
>
> 
>
> -----Ursprüngliche Nachricht-----
> Von: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von freeswitch@dalethatcher.com
> Gesendet: Samstag, 26. Januar 2008 16:30
> An: freeswitch-users@lists.freeswitch.org
> Betreff: [Freeswitch-users] Audio rate changes and clipping on playback
>
> Hello all!
>
> I'm experiencing some problems with clipping and the rate of playback of
> wav files from Freeswitch.  I've not raised this before as I thought the
> problems were due to hosting over an ADSL line.  However I've now seen
> the same issues running from a dedicated host.
>
> My test setup is:
>
>     Client <-> NAT ADSL <-> Freeswitch
>
> I've replicated using both the Gizmo and FWD clients from Windows.   If I
> substitute Asterisk with Freeswitch then the audio comes over fine.   The
> two test samples are one of myself and the fpm-sunshine.wav that is
> bundled with asterisk.
>
> I've tried playing around with the codecs but see the same problems
> regardless.  Also I've performed a test with a conference hosted on the
> Freeswitch instance with myself and friend dialling in using Gizmo and
> the audio is fine.  Which implied to me that the link between Gizmo to
> Freeswitch is ok.
>
> So I'm guessing that there is something wrong with the samples that is
> causing the problem?  They're both in this format:
>
> $ file fpm-sunshine.wav
> RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz
>
> My deployment platform is Debian stable and the Freeswitch SVN revision
> is 7343.  I've tried looking in Jira but keep getting a system error
> page (for search terms 'audio', 'wav' and others).
>
> Is anyone else seeing the same issues or has a solution?
>
> thanks,
>
> - Dale
>
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