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 <[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 > > _______________________________________________ -- 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