I'm running a TDM-400P with 2 x FXS and 2 x FXO. I'm finding that
there seems to be an odd relationship to sound quality on the card to my local
when connecting via a SIP client.
When I'm on my local network, if I connect to Asterisk via a SIP client
(such as x-pro), and dial an outside line through the card, sound quality seems
quite good.
However, when I'm at a remote location and connect via the same SIP client
and dial an outside line, the audio quality is fuzzy, sometimes quiet, and
generally more difficult to understand.
I spent a bunch of time troubleshooting the SIP end of things, thinking
that's where the problem was, until I realized that every other SIP
connection I make (from remote) yields a high quality call. ie. I can dial
another SIP client and maintain high quality audio. Additionally, I can dial an
extension that not only SIP connects to my server, but from there goes out an
IAX2 connection to another remote Asterisk server, from there to another SIP
client, and the audio quality is excellent.
Therefore, I don't think the audio issue I'm experiencing is on the SIP
end.
Are there some wierd SIP -> ZAP timing / conversion / other issues that could
be causing this?
thoughts?
regards,
Paul
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