Ya, I have done that, below is zapata.conf. Also we had an TMP card with analog lines. & SIP cals were great on them. & now when we switched over. SIP calls have echo.. which shouldnt be at all.<br><br>[channels]<br>language=en<br>#include zapata_additional.conf<br>context=from-pstn<br>switchtype=national<br>pridialplan=national<br>signalling=pri_cpe<br>faxdetect=incoming<br>usecallerid=yes<br>echocancel=yes<br>callerid=asreceived<br>echocancelwhenbridged=no<br>echotraining=128<br>;rxgain=-3.0<br>;txgain=-7.0<br>group=0<br>channel=1-23<br><br>--<br>Deepak<br><br><b><i>Alex Balashov <abalashov@evaristesys.com></i></b> wrote:<blockquote class="replbq" style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;"> On Sat, 9 Jun 2007, Deepak Naidu wrote:<br><br>> But now when we are live, there is a terrible echo between 2 SIP calls. <br>> If I call the same extension from outside the voice is clear.<br><br> My impression is
that the transcoding that takes place between two<br>purely software SIP calls never goes through the TE212P card.<br><br> There are probably echo cancellation options you can enable that are<br>relevant to software channels. I distantly recall there even being some<br>stuff youc an uncomment in the source.<br><br>> Totally lost Digium says we need to remove the echo module to resolve <br>> SIP echo problems. Then ? the heck we pay for...<br><br> Not sure why Digium would say that.<br><br>--<br>Alex Balashov<br>Evariste Systems<br>Web : http://www.evaristesys.com/<br>Tel : +1-678-954-0670<br>Direct : +1-678-954-0671<br>_______________________________________________<br>--Bandwidth and Colocation provided by Easynews.com --<br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> http://lists.digium.com/mailman/listinfo/asterisk-users<br></blockquote><br><p> 
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