Interesting news... I just got a call from one of the SIP phones outside
our LAN, over a VPN, with reinvite disabled, and it sounded like a
robot. Calls from SIP phones on the VPN sound fine when reinvite is
enabled. So it seems ANY call Asterisk bridges to the Polycom sounds
crappy.
Maybe this will shed some light on the issue.
Eric
Noah Miller wrote:
There's no transcoding going on. It's ulaw on IAX with Sixtel and ulaw
on SIP to the phone. I considered that as a possibility originally, and
even tried using GSM with Sixtel to force it to do transcoding, but had
the exact same problem.
The Asterisk box is a 2.4ghz P4 with 512MB RAM, doing nothing but
Asterisk. I have only 9 extensions.
I would think there's a possibility of packet loss on the IAX channel,
except the other SIP phones (SJPhone softphone) work flawlessly. Also,
OUTBOUND calls are just fine on the Polycoms. Only incoming calls are
messed up.
Just to cover all the bases, have you tried any other IAX providers or
connections?
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