On Dec 22, 2006, at 3:56 AM, Tony Mountifield wrote:
In article <[EMAIL PROTECTED]>,
Robert Jenkins <[EMAIL PROTECTED]> 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: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - 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
iaxtel: 700:248:2637 x:1500
http://www.sheltonjohns.com
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