[Asterisk-Users] Asterisk 1.00 Call quality problem

[Nick Cobley]

I upgraded from RC2 last night, but have a major call quality issue.
Heres our setup:

1 FXS and 1 FXO card.
Incoming/Outgoing calls via IAX trunking from our provider. G729
running between us and the VoIP provider.
Two handsets, one BudgetTone 102 and a Cisco 7940G running the  7.2
SIP firmware.
Both these phones are using ULAW to the server, and we have plenty of
G729 licenses on the server.

Now the BudgetTone does not have any problems at all. The Cisco
however has severe breakup on the incoming audio from the VoIP
provider. The caller can hear me fine. In some cases the call is okay
for 10seconds or so then slowly gets worse, but most of the time I
just cant make out what they are saying.

The link we are running this on is a 1500/256 ADSL line, we are not
running much else on this link so it does not appear to be bandwidth
related.

I tried changing codecs that the phone uses, in particular to G729 so
there is no conversion taking place, but it does not make any
difference at all. I just dont understand why it only affects the
Cisco.

I just switched it back to RC2 and it works fine again. This phone
does not have any problems at all when taking/making calls over the
Zaptel interface.

Any ideas?