[Asterisk-Users] Random Intermittent Noise for SIP to FX0 calls plus echo

[Rich Adamson]

> 	Was wondering if anyone out there has had the experience I have been
> having. In reading recent posts on echo cancellation, I think there
> is....
> 
> 	We recently cut over the Asterisk and are configured with 5 FXS and 2
> FXO ports to the PSTN via 2 TDM400P's and 5 SIP phones on our local
> network. I have set up echo cancellation with 800ms echo training. I do
> not have echocancelwhenbridged on, since since this is for complete TDM
> circuit per the comments in zapata.conf. When calls come in, there is
> echo, but it quickly trains and goes away. This is not the problem
> though. The asterisk server has 1GB RAM and 1GHz clock. We are currently
> using u-law codecs only. Digium support said that I might want to play
> around with using other codecs and the RX/TX gain to see if that makes a
> difference. These gain settings are not set and therefore taking the
> default. Before I start shooting in the dark, I thought I'd go to this
> group to see if the following problem has been solved before I start
> shooting in the dark.
> 
> 	The problem we are having is that every now and then, say 4 times over
> a 20 minute call, interference occurs in the ear of the SIP phone user.
> The other side (PSTN caller) of the conversation may hear a few short
> (half second) breaks in the conversation. The characteristics of the
> interference in the SIP phone user is some in-band MF tones which may
> start out faint and get louder which are broken up them selves to make a
> cracking noise. There may be some momentary echo of my voice when this
> happens also. This may last for about 3 seconds and then the
> conversation is clear.
> 
> 	We don't get this when using Analog phones on an FXS port nor for SIP
> to SIP conversations.
> 
> Has anyone else experienced this and resolved it?

Sounds sort of like a possible issue with the ethernet. Are you sure
you're running full-duplex on * and the switch its attached to, etc?