[asterisk-users] Half-duplex call on TDM2400p with VPMADT032
Ruben Zamora
ruben.zamora at zys.com.mx
Mon Apr 7 21:22:30 CDT 2008
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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>
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