[Asterisk-Users] Sound Quality.

[Michael Baird]

I've been using asterisk for a while, only for dialout from a SIP client
over a PRI -> PSTN, this works great. Now I have a need to also dialin
to asterisk over the PRI/TDM, I've been testing by creating an
extension, and essentially playing back a recording on that extension.
If I access the extension from a SIP client, it works great, very high
quality, no chops are stumbles, If I come into the same extension from
the PSTN over the PRI into asterisk, the quality degrades significantly,
I can hear some noise on answer, and flutters while the message is
played. I'm wondering if there are some settings I should try to
increase the quality (other then the gain and echo settings in
zapata.conf).

Regards
MIKE