[Asterisk-Users] Choppy Sound on PSTN End

[Aza]

Hi,

I have the same problem on a Dell 1850 with a TE410P, static/chop on calls
to through the TE410P, and have been attempting to narrow it down for the
last week. Interrupts don't seem to be a problem and I have two PRIs from
two different suppliers and both have the same static/chop on the line so
it's not the PRI.

The leading suspect at the moment is the RAID controller. Unfortunately it's
rather difficult to remove this from the set up but I plan to switch one of
the PRIs to a Dell 1750 without a RAID controller to see if the problem
still goes away.

Aaron

________________________________________
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Tim Chandler
Sent: 02 May 2005 17:23
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Choppy Sound on PSTN End

Hi all,
I recently set up Asterisk on a Dell PowerEdge 1850 Dual Xeon 2.8 Ghz
processor.  I am running the latest build of White Box Enterprise Linux.
Our call routing is like this:
SJPHONE on Windows -> QoS-enabled Switch -> Asterisk -> T1 Line ->
Broadvoice SIP account -> PSTN
Calls seem to work great from user to user.  However, calls from a SJPhone
user to the PSTN are not so great.  The SJPhone user hears the person on the
PSTN perfectly – I mean, completely flawless.  However, the user on the PSTN
end hears choppy / jittery, extraneous clicks, etc.
Here is the SJPhone config:
Audio Compression: G.711
Driver buffer size: 20 msec
Driver input queue length: 6
Driver output queue length: 4
RTP jitter queue length: 6
Silence Suppression: No
DTMF Sending: RFC 2833
Signal Duration (ms): 270
RTP Payload type: 101
Signal volume: 10
Pause duration (ms): 100

And the sip extension config (in Asterisk Management Portal):
Allow: blank
Canreinvite: no
Disallow: gsm
Dtmfmode: rfc2833
Host: dynamic
Nat: yes (some users are behind NAT)
Qualify: no

Any ideas on what to do to get rid of the choppiness?
Thanks!
Tim