asterisk users - Jul 2003 - Sound Quality.

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

Hi I'm getting really bad static on forwarded calls to the point of 
not being able to hear the person at the other end. I'm running an E1 
line in and everything else is fine. I'm also getting this error:
Sep 11 14:56:56 WARNING[18295]: chan_sip.c:2561 sip_write: Asked to 
transmit frame type 64, while native formats is 4 (read/write = 64/64)
 
I'm running Asterisk 1.2.11 and have tried a couple of different codecs 
in SIP.conf.
Any ideas??

Cheers,

 Joel Hill
Support Engineer
Asterisk IT
www.asteriskit.com.au
www.theasteriskshop.com.au