[Asterisk-Users] Poor quality with FWD - codec selection issue?

[Iain Stevenson]

A colleague called me through my * system via FWD using SJPhone and the 
quality was distinctly poor - a lot of hum and delay.  Looking at the debug 
log the codec used was miscellaneously numbered 0, 4 and 8.  I thought I'd 
disabled 4 (g.723) but it appears not.  My sip.conf has this:

general]
port = 5060                     ; Port to bind to
bindaddr = 0.0.0.0              ; Address to bind to
context = voip-sip
defaultexpiry = 3600
register => 12345:secret at fwd.pulver.com/39
disallow=all
allow=alaw
allow=ulaw

I was expecting this would stop g.723 from being even tried - am I missing 
something?

Is there any config option for SJphone that clobbers g.723?

  Iain