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<DIV><FONT face=Arial size=2>I am fighitng with this problem since last
week.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>We use sipura 2100 ATA configured with rtp lenght
about 20ms.</FONT></DIV>
<DIV><FONT face=Arial size=2>Asterisk is connected to our upstrim using pri
(Sangoma aft104d)</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>During the call the pstn side hear a lot of ticks,
I changed all kind of jitter buffer into the ata and patched asterisk to have jb
on sip too......but the issue has not been fixed.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I am pretty sure that the ata are fine becuase If I
use a Mediatrix 2631 instead of asterisk, the audio quality is
perfect.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>So I am little bit lost about it.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Any suggestion?</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Giving the fact the problem is present only on the
pstn side it has to do with the rtp stream exiting the ata or entring the
asterisk box.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Regards</FONT></DIV></BODY></HTML>