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<DIV dir=ltr align=left><SPAN class=164511917-02052005><FONT face=Arial
color=#0000ff size=2> I have the exact setup you describe,
SJPhone -> * -> Zap/PRI. I think you need to twiddle some settings. You
might turn on qualify just to see if the * is seeing network flaws. Keep in
mind, if your using windows, anytime the user starts clicking around, you can
expect less than ideal audio. Also, why disable GSM ?</FONT></SPAN></DIV><BR>
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<FONT face=Tahoma size=2><B>From:</B> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <B>On Behalf Of </B>Tim
Chandler<BR><B>Sent:</B> Monday, May 02, 2005 11:23 AM<BR><B>To:</B>
asterisk-users@lists.digium.com<BR><B>Subject:</B> [Asterisk-Users] Choppy Sound
on PSTN End<BR></FONT><BR></DIV>
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<P align=left><SPAN lang=en-us><FONT face=Arial size=2>Hi all,</FONT></SPAN></P>
<P align=left><SPAN lang=en-us><FONT face=Arial size=2>I recently set up
Asterisk on a Dell PowerEdge 1850 Dual Xeon 2.8 Ghz processor. I am
running</FONT></SPAN><SPAN lang=en-us></SPAN><SPAN lang=en-us> <FONT face=Arial
size=2>the latest build of</FONT></SPAN><SPAN lang=en-us></SPAN><SPAN
lang=en-us> <FONT face=Arial size=2>White Box Enterprise
Linux</FONT></SPAN><SPAN lang=en-us></SPAN><SPAN lang=en-us><FONT face=Arial
size=2>.</FONT></SPAN></P>
<P align=left><SPAN lang=en-us><FONT face=Arial size=2>Our call routing is like
this:</FONT></SPAN></P>
<P align=left><SPAN lang=en-us><FONT face=Arial size=2>SJPHONE on Windows ->
QoS-enabled Switch -> Asterisk -> T1 Line -> Broadvoice SIP
account</FONT></SPAN><SPAN lang=en-us></SPAN><SPAN lang=en-us><FONT face=Arial
size=2> -> PSTN</FONT></SPAN><SPAN lang=en-us></SPAN><SPAN
lang=en-us></SPAN></P>
<P align=left><SPAN lang=en-us><FONT face=Arial size=2>Calls seem to work great
from user to user. However, calls fro</FONT></SPAN><SPAN
lang=en-us></SPAN><SPAN lang=en-us><FONT face=Arial size=2>m a SJPhone user to
the PSTN are not so great. The SJPhone user hears the person on the PSTN
perfectly</FONT></SPAN><SPAN lang=en-us></SPAN><SPAN lang=en-us> <FONT
face=Arial size=2>–</FONT></SPAN><SPAN lang=en-us></SPAN><SPAN lang=en-us><FONT
face=Arial size=2> I mean, completely flawless. However, the user on the
PSTN end hears choppy / jittery, extraneous clicks, etc.</FONT></SPAN></P>
<P align=left><SPAN lang=en-us><FONT face=Arial size=2>Here is the SJPhone
config:</FONT></SPAN></P>
<P align=left><SPAN lang=en-us><FONT face=Arial size=2>Audio Compression:
G.711</FONT></SPAN></P>
<P align=left><SPAN lang=en-us><FONT face=Arial size=2>Driver buffer size: 20
msec</FONT></SPAN></P>
<P align=left><SPAN lang=en-us><FONT face=Arial size=2>Driver input queue
length: 6</FONT></SPAN></P>
<P align=left><SPAN lang=en-us><FONT face=Arial size=2>Driver</FONT></SPAN><SPAN
lang=en-us></SPAN><SPAN lang=en-us> <FONT face=Arial
size=2>output</FONT></SPAN><SPAN lang=en-us></SPAN><SPAN lang=en-us> <FONT
face=Arial size=2>queue length:</FONT></SPAN><SPAN lang=en-us></SPAN><SPAN
lang=en-us> <FONT face=Arial size=2>4</FONT></SPAN></P>
<P align=left><SPAN lang=en-us><FONT face=Arial size=2>RTP jitter queue length:
6</FONT></SPAN></P>
<P align=left><SPAN lang=en-us><FONT face=Arial
size=2>Silence</FONT></SPAN><SPAN lang=en-us></SPAN><SPAN lang=en-us> <FONT
face=Arial size=2>Suppression:</FONT></SPAN><SPAN lang=en-us></SPAN><SPAN
lang=en-us><FONT face=Arial size=2> No</FONT></SPAN><SPAN
lang=en-us></SPAN><SPAN lang=en-us></SPAN></P>
<P align=left><SPAN lang=en-us><FONT face=Arial size=2>DTMF Sending: RFC
2833</FONT></SPAN></P>
<P align=left><SPAN lang=en-us><FONT face=Arial size=2>Signal Duration (ms):
270</FONT></SPAN></P>
<P align=left><SPAN lang=en-us><FONT face=Arial size=2>RTP Payload type:
101</FONT></SPAN></P>
<P align=left><SPAN lang=en-us><FONT face=Arial size=2>Signal volume:
10</FONT></SPAN></P>
<P align=left><SPAN lang=en-us><FONT face=Arial size=2>Pause duration (ms):
100</FONT></SPAN></P><BR>
<P align=left><SPAN lang=en-us><FONT face=Arial size=2>And the sip extension
config (in Asterisk Management Portal):</FONT></SPAN></P>
<P align=left><SPAN lang=en-us><FONT face=Arial size=2>Allow:
blank</FONT></SPAN></P>
<P align=left><SPAN lang=en-us><FONT face=Arial size=2>Canreinvite:
no</FONT></SPAN></P>
<P align=left><SPAN lang=en-us><FONT face=Arial size=2>Disallow:
gsm</FONT></SPAN></P>
<P align=left><SPAN lang=en-us><FONT face=Arial size=2>Dtmfmode:
rfc2833</FONT></SPAN></P>
<P align=left><SPAN lang=en-us><FONT face=Arial size=2>Host:
dynamic</FONT></SPAN></P>
<P align=left><SPAN lang=en-us><FONT face=Arial size=2>Nat: yes (some users are
behind NAT)</FONT></SPAN><SPAN lang=en-us></SPAN><SPAN lang=en-us></SPAN></P>
<P align=left><SPAN lang=en-us><FONT face=Arial size=2>Qualify:
no</FONT></SPAN></P><BR>
<P align=left><SPAN lang=en-us><FONT face=Arial size=2>Any ideas on what to do
to get rid of the choppiness?</FONT></SPAN></P>
<P align=left><SPAN lang=en-us><FONT face=Arial size=2>Thanks!</FONT></SPAN></P>
<P align=left><SPAN lang=en-us><FONT face=Arial size=2>Tim</FONT></SPAN><SPAN
lang=en-us></SPAN><SPAN lang=en-us></SPAN></P></BODY></HTML>