[asterisk-users] Polycom 601 Issues

[Nathan Alberti]

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.