[Asterisk-Users] Call quality problems

[Michael Welter]

I'm having difficulty with an Asterisk system.  The external party has 
very good call quality, but the internal party hears clipping and drop outs.

The WAN comes in from the Cisco IAD and into a LAN switch (DLink 
DGS-1005D w/ 802.1p) where the two public IPs are switched to different 
devices.  One device is a FireBox device controlling a separate LAN with 
VPNs.  The other device is eth0 on the Asterisk system.

On the Asterisk eth1 is a 3Com 2226 LAN switch which connects Polycom 
IP501 phones.  There are no PCs on this voice LAN.  All ports on all LAN 
switches indicate full duplex.  The quality problem doesn't appear to be 
volume related (a single call still has problems).

The Polycom IP501s use SIP to the PBX, and the PBX uses SIP to the provider.

The normal WWV time signal consists of a constant tone that is 
interrupted every second by a click.  On the Polycom, each click can be 
heard, the tone starts, but the tone is clipped and there is silence 
until the next click.

I've verified that QoS is enabled in the IAD.

I would appreciate your thoughts.

Thanks,

-- 
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
mike at TelecomMatters.net
www.TelecomMatters.net