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

[Lex Lethol]

Ruben,

Contact support at digium they have a release on a firmware that fixes
this and other issues with the VPMADT032.

Apparently it comes on newer zaptel drivers.

Good luck with your install.

Lex

On Mon, Apr 7, 2008 at 3:11 PM, Matthew Fredrickson <creslin at digium.com> wrote:
> Ruben Zamora wrote:
>  > 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.
>
>  Please contact Digium technical support about this.  This is definitely
>  something that we need to work with the vendor of the echo canceller IP
>  about.
>
>  Matthew Fredrickson
>
>
>
>  >
>  > Thanks
>  > Ruben
>  >
>  > Lex Lethol escribió:
>  >> 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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>
>  --
>  Matthew Fredrickson
>  Software/Firmware Engineer
>  Digium, Inc.
>
>
>
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