asterisk users - Nov 2005 - 7960 audio quality when calling remote asterisk box

Hello all,

I've been doing some testing with the 7960s I have here calling into a
remote asterisk box (1.0.9). Audio quality on the 7960 is perfect when I
call to other extensions on my local asterisk (1.2.0), but when I place
calls to users on the remote box (boxes are linked via IAX2) audio quality
drops massively - the party at the other end can hear what I'm saying
perfectly, but I can barely make out one word in three.

I then tried the same thing using a sip phone, and the audio problems aren't
there at all.

To summarize:
audio problems: 7960 -> local asterisk (1.2) -> remote asterisk (1.0.9)
->
sip phone
no problems: sip phone -> local asterisk (1.2) -> remote asterisk (1.0.9)
->
sip phone
no problems: 7960 -> local asterisk (1.2) -> sip phone
no problems: 7960 -> local asterisk (1.2) -> pstn

I've tried disabling the IAX2 jitter buffer on both asterisks and forcing
both of them to use the same codec, all without success.

I'd be grateful for any hints as to which options I should check.

Thanks in advance.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
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