On 18/08/2006, at 10:43 PM, Noah Miller wrote:
Hi Nathan -
The problem occurs during transfer and hold retrieval, answering the
call is fine, the call is put on hold then either a transfer is
attempted or the call is retrieved from hold. When this is attempted
the remote party (i.e. the caller in the case of a hold retrieval)
cannot hear the receptionist at all for the first few seconds, then
slowly they are able to hear fragments of the voice which is
basically stuttered and robotic.
Unfortunately I cannot replicate this in my office, the only
difference between the two configurations is the model of Cisco
switch but I really don't think this should be making any difference.
Do you get any CLI errors or log messages that might tell us anything
more? I too have Cisco switches and Polycom 601's at one location and
I don't have these problems (I'm using 1.2.10). What codec are they
using?
You can do a LOT to a Cisco switch to make it handle traffic in
different ways. Do you have access to their switch to see if they've
done any prioritization of traffic or have any other "unusual"
settings?
No, absolutely no CLI errors at all, the calls are re-inviting and
using G711A.
There are no errors on the port at all, I do have access to the
switch and it really has no special configuration for traffic
prioritization.
As the RTP audio is flowing directly between the Cisco router (acting
as a media gateway) I am thinking its probably not an asterisk
problem so maybe I need to look at the configuration of the Cisco's
dial-peer.
Regards,
Nathan.
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