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<p class=MsoNormal><font size=2 face=Arial><span lang=EN-GB style='font-size:
10.0pt;font-family:Arial'>Hi All,<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-GB style='font-size:
10.0pt;font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-GB style='font-size:
10.0pt;font-family:Arial'>I’ve a 1.4.15 A*k server supporting several
users (approx 80 total, but <10 sim calls usually). I’ve one
user who complains of intermittent bad calls, though I suspect the bad calls
are across the board, but intermittent.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-GB style='font-size:
10.0pt;font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-GB style='font-size:
10.0pt;font-family:Arial'>Inbound calls are via in IAX trunk from <st1:PersonName
w:st="on">Gradwell</st1:PersonName>. CPU stats say that Asterisk never uses
more than 4-5% cpu, systems idle besides that. Memory seems ok too. Network
utilisation is < 300kbps. The voice network (clients + server) sit on
their own dedicated 100Mb switches. Stats from the switch say its lightly
loaded.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-GB style='font-size:
10.0pt;font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-GB style='font-size:
10.0pt;font-family:Arial'>I’ve turned on voicefile recording. What
we hear, when there is a bad call, is stuttered speech, from BOTH sides (so
local SIP client, and remote IAX inbound call).<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-GB style='font-size:
10.0pt;font-family:Arial'>Debug from asterisk just shows the call inbound,
answered and then hung up as per normal.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-GB style='font-size:
10.0pt;font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-GB style='font-size:
10.0pt;font-family:Arial'>I’m at a loss of how to debug the voice issue
further, without putting a wireshark PC on the switch, port-mirroring the
server and then capturing all of the traffic in a round-robin-type capture and
even then I’m not sure what that will achieve.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-GB style='font-size:
10.0pt;font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-GB style='font-size:
10.0pt;font-family:Arial'>I’m going to switch from IAX to SIP for the
inbound calls for that user and see if that helps.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-GB style='font-size:
10.0pt;font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-GB style='font-size:
10.0pt;font-family:Arial'>Any ideas welcome,<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-GB style='font-size:
10.0pt;font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-GB style='font-size:
10.0pt;font-family:Arial'>Thanks<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-GB style='font-size:
10.0pt;font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><st1:City w:st="on"><st1:place w:st="on"><font size=2
face=Arial><span lang=EN-GB style='font-size:10.0pt;font-family:Arial'>Adrian</span></font></st1:place></st1:City><font
size=2 face=Arial><span lang=EN-GB style='font-size:10.0pt;font-family:Arial'><o:p></o:p></span></font></p>
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