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. 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
