asterisk users - Jun 2004 - Eliminating silence suppression(?) on IAX2 calls

We have an Asterisk server that speaks IAX2 to Magrathea to get to the
PSTN. Our local phones are a mix of Cisco 7940s and Grandstream BT100s
all configured for SIP with silence-suppression disabled. Everything
is configured to use a-law encoding. The version is:

sip*CLI> show version
Asterisk CVS-05/06/04-18:45:57 built by root@sip on a i686 running Linux

Incoming callers are complaining of calls sounding a bit odd. With a
bit of experimentation, it seems that there's some form of silence
suppression cutting in that clips the "sides" of words. This
doesn't
happen on internal calls.

Since the problem is limited to external calls, I take it that the
problem relates to IAX2 and not some other part of Asterisk? Here's
a sanitised version of iax.conf:

[general]
port=5036
bandwidth=low
disallow=all
allow=alaw
allow=gsm
disallow=lpc10                  ; Icky sound quality...  Mr. Roboto.
jitterbuffer=no

register => USERNAME:PASSWORD@voip-gw1.magrathea-telecom.co.uk

tos=lowdelay

[magrathea]
type=friend
trunk=yes
host=dynamic
auth=plaintext
username=USERNAME
secret=PASSWORD
context=inbound-magrathea
notransfer=yes

Does anybody have any suggestions as to how to clear this odd audio
issue, or pointers to what configuration options I should tweak?
Ideally, I just want a completely uncompressed 64k channel, as if we
were just using ISDN.

Thanks in advance.

-- 
What contemptible scoundrel has stolen the cork to my lunch?
								- W.C. Fields

>
>[general]
>port=5036
>bandwidth=low
>disallow=all
>allow=alaw
>allow=gsm
>disallow=lpc10                  ; Icky sound quality...  Mr. Roboto.
>jitterbuffer=no
>  
>
Try turning the jitterbuffer on and see if it helps.

-- 
Andres
Network Admin
http://www.telesip.net