[asterisk-users] Half-duplex call on TDM2400p with VPMADT032

[Lex Lethol]

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