|mkn0014 at gmail.com
|Posted: Tue Jan 01, 2008 2:38 am Post subject: [asterisk-users] One Way Delay in Audio Over Analog
|Brian Alexander wrote:
|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!
What about some facts ?
Software versions ?