asterisk users - Apr 2008 - Half-duplex call on TDM2400p with VPMADT032

Hi,

I've used all kinds of digium cards without troubles.  My last
installation is using a TDM2400p with VPMADT032 echo cancel module and
after a week of use we noticed that any incoming audio stream gets
clipped / dropped when you speak or when ambient noise is high.  The
call basically feels as in a half-duplex channel, but only to the
person behind our asterisk.  I found a quick way to recreate by
placing a call using zapata channel, someplace that has an audio
stream (ie. music on hold from another pbx).  When one talks into the
phone, one can notice the incoming audio getting muted until you stop
talking.

First I thought it had to do with polycom configuration although we
use the same setup for all installations (VAD, etc), but the same
happens with other sip phones and after more tests I can only recreate
this using the TDM2400p's FXO trunks.  I have an older TDM2400p with
no VPMADT032 in production (without this problem), this leads me to
believe there maybe something wrong with VPMADT032 module or with my
card in particular.

Today I rebuilt everything from scratch using latest asterisk 1.2
release, rechecked with the TDM2400p manual zapata configs just to
make sure I wasn't missing something.  As the manual suggests, I am
just using echocancel=yes and this should set 128 default value for
the card.  In the general zapata options there we have
echocancelwhenbridged=yes.  I have played with all yes/no combinations
without luck.

Interrupts and timing stuff are OK, we have good incoming and outgoing
audio quality (as long as its not at the same time).

Anyone else using this card showing the same problems?

Any zaptel/asterisk gurus wanna take a shot at this?

Thanks in advance for your feedback/comments.

Lex

Hi,
I have a same problem, last week i was working with TE120 with a little 
echo in some call,  I replace the card
with a TE122B ( Included Echo Cancelation VPMADT032) and there was no 
more echo in my call.

But know i have de same probelm with my incoming audio stream gets 
clipped / dropped when you speak.

Thanks
Ruben

Lex Lethol escribi?:
> Hi,
>
> I've used all kinds of digium cards without troubles.  My last
> installation is using a TDM2400p with VPMADT032 echo cancel module and
> after a week of use we noticed that any incoming audio stream gets
> clipped / dropped when you speak or when ambient noise is high.  The
> call basically feels as in a half-duplex channel, but only to the
> person behind our asterisk.  I found a quick way to recreate by
> placing a call using zapata channel, someplace that has an audio
> stream (ie. music on hold from another pbx).  When one talks into the
> phone, one can notice the incoming audio getting muted until you stop
> talking.
>
> First I thought it had to do with polycom configuration although we
> use the same setup for all installations (VAD, etc), but the same
> happens with other sip phones and after more tests I can only recreate
> this using the TDM2400p's FXO trunks.  I have an older TDM2400p with
> no VPMADT032 in production (without this problem), this leads me to
> believe there maybe something wrong with VPMADT032 module or with my
> card in particular.
>
> Today I rebuilt everything from scratch using latest asterisk 1.2
> release, rechecked with the TDM2400p manual zapata configs just to
> make sure I wasn't missing something.  As the manual suggests, I am
> just using echocancel=yes and this should set 128 default value for
> the card.  In the general zapata options there we have
> echocancelwhenbridged=yes.  I have played with all yes/no combinations
> without luck.
>
> Interrupts and timing stuff are OK, we have good incoming and outgoing
> audio quality (as long as its not at the same time).
>
> Anyone else using this card showing the same problems?
>
> Any zaptel/asterisk gurus wanna take a shot at this?
>
> Thanks in advance for your feedback/comments.
>
> Lex
>
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