asterisk users - Sep 2004 - Random Intermittent Noise for SIP to FX0 calls plus echo

Dear group,

	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?


Thanks,
Mike Meyer

Mike Meyer wrote:
> 	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.
> 
Exact same issues here, haven't solved yet, but am slowly working on it.

Thanks,
Chris

> 	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?