[Asterisk-Users] Asterisk 1.00 Call quality problem

[steve at daviesfam.org]

On Thu, 30 Sep 2004, Nick Cobley wrote:

> 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.


Make sure you aren't trying to use trunking and the iax jitter buffer 
together.  As of now they don't work properly together.  So turn off one 
or the other for the link to your VOIP provider.

As at 1.0 the jitter buffer can be enabled and disabled per peer/user.  
See iax.conf.sample.

Steve