[asterisk-users] Re: Help with silence or gating of speech?

[Bryan M. Johns]

On Dec 22, 2006, at 3:56 AM, Tony Mountifield wrote:

> In article <0ed301c7255b$542a2bd0$2800a8c0 at office>,
> Robert Jenkins <raj at jrw.co.uk> wrote:
>> Hi,
>>
>> I'm using Asterisk (1.2.13) on Centos 4.4 x86_64 with a TDM2400E  
>> for analog
>> trunks (& extensions) plus some Polycom 501 & 601 phones.
>>
>> I have a problem in that the audio via the Polycoms is gated or  
>> muted during
>> quiet parts of the other person's speech.
>
> I'm not familiar with the Polycoms, but I would guess that they  
> have some
> kind of web-based setup screens. Look for "silence suppression" or  
> "VAD
> (voice activity detection)" and set it to Off or Disabled.
>
> Otherwise you could try visiting http://bugs.digium.com/view.php? 
> id=5374
> and apply the patch 2005-10-04-3-asynchronous.patch
>
> It's the patch to channel.c that is the important part.
>
> Cheers
> Tony
> -- 
> Tony Mountifield
> Work: tony at softins.co.uk - http://www.softins.co.uk
> Play: tony at mountifield.org - http://tony.mountifield.org
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The PolyCom handsets have lots of tune-able audio parameters that are  
manipulated using the phone's configuration file.  Assuming that you  
are using TFTP or FTP config distribution, these files are the  
[phonemacaddress].cfg file and the master sip.cfg file.

I recommend looking through the settings available in these files and  
maybe doing some research on the config of particular PolyCom models  
at http://www.voip-info.org.  Backup your original configs before  
making edits, though ;-).

I hope this is helpful.

Bryan M. Johns
Partner
Shelton | Johns Technology Group
office: 678:248:2637 x:1500
direct: 678:229:1809
mobile: 404.259.9216
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http://www.sheltonjohns.com


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