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 <[EMAIL PROTECTED]> 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 > >> > >> _______________________________________________ > >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > >> > >> asterisk-users mailing list > >> To UNSUBSCRIBE or update options visit: > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> > >> > > > > _______________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > Matthew Fredrickson > Software/Firmware Engineer > Digium, Inc. > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
