Re: [asterisk-users] fxotune failure! Could not fill input buffer - got -1 bytes, expected 4000 bytes

2006-09-12 Thread Jeff Turner
On Wed, Sep 13, 2006 at 12:27:48AM -0400, Jonathan Barratt wrote:
 I'm using Zaptel 1.2.9.1 and receiving the error mentioned in the title when
 doing an fxotune -i on a TDM400p, no matter what other parameters I try.
 
 Has anyone seen this problem and solved it?

Not sure what causes that, but try prepending 'strace' to the command and
see what it's trying to do when it fails.


--Jeff

 Please, please, please help.
 
 Fxotune has worked for me in the past and I desperately need to clear up the
 echo in this office.
 
 Thanks in advance,
 Jonathan
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Re: [asterisk-users] Re: Really bad phone line.. possible causes?

2006-09-07 Thread Jeff Turner
On Thu, Sep 07, 2006 at 07:50:39AM -0800, Mojo with Horan  Company, LLC wrote:
 It is in zconfig.h -- immediately before the echo cans:
 /* #define CONFIG_ZAPTEL_MMX */
 Just make sure it's still commented out to give my situation a try.

Yes, it's commented out. I'm not sure what codec these Polycom phones use
(there's nothing codec-related in sip.conf). I'll run asterisk in debug
mode and find out.


Cheers,
Jeff

 Moj
 
 M.Hockings wrote:
  Mojo with Horan  Company, LLC wrote:
  What codec are your sip phones using?  We'd have a similar, though 
  immediate, degradation in audio quality using G.729 when zaptel was 
  built with MMX optimizations.  We use an AMD CPU.
 
  When zaptel was rebuilt without MMX optimizations we were back in business.
 
  
  How do you configure it sans MMX ?
  
  I had a similar problem and changed the echo can from the default to the 
one where the comments in the .h file say to try if the first one does 
  not work (sorry can't remember it's name offhand).  It was like night 
  and day, works fine now.
  
  Mike
  
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 -- 
 Mojo [EMAIL PROTECTED]
 Office Manager, Horan  Company, LLC
 (907) 747- x112
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[asterisk-users] Really bad phone line.. possible causes?

2006-09-05 Thread Jeff Turner
Hi,

I was wondering if those more familiar with PSTN-Asterisk installs
could take a listen to this (86k recording):

http://confluence.atlassian.com/download/attachments/185668/linegoingbad.mp3?version=1

It's what I hear dialling into our Asterisk box. As soon as the call
receiver makes a sound (clicking fingers in that recording), the line
makes a brief buzzing noise, then goes crazy with raw static.

There is a longer recording and configuration info at:

http://confluence.atlassian.com/display/TEST/Asterisk+phoneline+samples

The odd thing is, we have two phonelines, and the connection quality of
the second line coming into Asterisk is perfect.

Our Asterisk box is hooked up (in parallel with a NEC PBX) to our public
phone line via a TDM404P card.

Does this ring any bells for anyone?


Thanks,
Jeff
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[asterisk-users] [OT] FYI: Polycom phone intermittent disconnects

2006-08-02 Thread Jeff Turner
Just a note for Polycom phone users, that will hopefully help someone.

Ever since deploying an office full of Polycom 601 phones, some users
have experienced intermittent disconnects, where voice transmit dies, or
both receive and transmit dies. Absolutely nothing in the Asterisk logs.

Solution: plug the socket into the handset in properly! Pushing the
socket in, it make a nice 'click' and _seems_ to be in, but it's not (and
is a bit wobbly). Push it further, until the plastic hook is not exposed
at all, and it makes another click. Now it's in :)


--Jeff
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