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

[Ruben Zamora]

Lex

Thanks a lot.   These morning i call Digium Support.   One issue that i 
miss in my before e-mail is that i have
my Asterisk installed in Monterrey,Mexico and my TELCO provider gave my 
MFC/R2. 
Asterisk 1.4.18,Zaptel 1.4.9.2 and Unicall.

They told me they can help me because they dont have UNICALL support.

So... I need to investigate more or wait for a new zaptel or anything else.

By the moment i have a big problem.

Thanks

Ruben




Lex Lethol escribió:
> 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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>>  >
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>>
>>  --
>>  Matthew Fredrickson
>>  Software/Firmware Engineer
>>  Digium, Inc.
>>
>>
>>
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