[asterisk-users] One Way Delay in Audio Over Analog

[shadowym]

What are you using for a PSTN gateway?
 
From: Brian Alexander [mailto:balexander at visionpointsystems.com] 
Sent: Monday, December 31, 2007 3:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] One Way Delay in Audio Over Analog
 
I have been trying to track down the cause/fix for a problem and I am out of
ideas... I am hoping one of you can point me in the right direction.

The symptom is that when a calls is placed from an internal extension
through an analog line to a number on the pstn the caller can hear the
callee but the callee can not hear the caller for as long as ten seconds. 

The problem appears to happen fairly consistently on the same pstn numbers.
However, I have not seen a common characteristic in those numbers. For
example, one of them is a direct number to a cell phone and another is to a
Verizon fiber-optic phone/data service. 

The problem does not seem to be related to the type of SIP phone being used
by the caller - for example, we have tried both X-Lite and Polycom phones
without a change in behavior.

The problem does not appear to occur if the callee then calls into our
system (at least the one time I was able to have this happen). 

Turning on or off echo cancellation and/or call progress does not seem to
change the behavior. 

I will appreciate any ideas you have. I am certainly stumped.

Thanks and Happy New Year!
-Brian
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