I am fighitng with this problem since last week. We use sipura 2100 ATA configured with rtp lenght about 20ms. Asterisk is connected to our upstrim using pri (Sangoma aft104d) 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. I am pretty sure that the ata are fine becuase If I use a Mediatrix 2631 instead of asterisk, the audio quality is perfect. So I am little bit lost about it. Any suggestion? 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. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060826/5925e289/attachment.htm