<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> </head> <body bgcolor="#ffffff" text="#000000"> <span class="postbody">Greetings,<br> I am running Asterisk 1.4.25 with Dahdi Complete 2.2.0, on a Digium TE121B PCI express card with a </span>VPMADT032<span class="postbody"> echo cancellation module, connected to an AT&T 24 channel PRI.</span><br> <br> When I run dahdi show channel X on an active channel, I see this:<br> <blockquote>Echo Cancellation: 128 taps unless TDM bridged, currently ON<br> </blockquote> So I know the echo cancellation is working, however when I call a local analog land line, I get discernible echo.<br> <br> Here is my chan_dahdi.conf:<br> <br> [channels]<br> ; configuration for T1 card as PRI<br> language = en<br> <br> group = 1<br> echocancel = yes<br> echotraining = yes<br> signalling = pri_cpe<br> switchtype = 4ess<br> usecallerid = yes<br> context = incoming<br> channel => 1-23<br> <br> Someone has to have had some experience with these hardware echo cancellers, any ideas? Should I adjust my rx and tx gains? Any advice would be very helpful. Thank you.<br> <div class="moz-signature">-- <br> <title></title> <p><font face="Arial, Helvetica, sans-serif"><b>Jason Baker<br> </b><font color="#660000"><span style=""><i>IT Coordinator</i></span></font></font></p> <font color="#000099" face="Arial, Helvetica, sans-serif"><b> Glastender, Inc.</b></font><br> <font size="2">5400 North Michigan Road<br> Saginaw, Michigan 48604 USA<br> Phone: 989.752.4275 ext. 228<br> Fax: 989.752.4276</font><br> <a href="http://www.glastender.com/">www.glastender.com</a> </div> </body> </html>
Jason Baker wrote:> So I know the echo cancellation is working, however when I call a > local analog land line, I get discernible echo. >echocancelwhenbridged=yes Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."
Jason Baker wrote:> > language = en > > group = 1 > echocancel = yes > echotraining = yes > signalling = pri_cpe > switchtype = 4ess > usecallerid = yes > context = incoming > channel => 1-23Just noted that your system is out of Saginaw. The system below is out of Livonia, with an AT&T PRI as well. Note the rx/txgain entries, it may be useful as well: switchtype=national context=pri signalling=pri_cpe echocancel=yes echotraining=yes echocancelwhenbridged=yes rxgain=-1.0 txgain=-4.0 -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."
No it's not a fact of life. VoIP works as fine as conventional telephony once it's correctly set up. Try echocancel=256 instead of echocancel=yes and also run fxotune (check the man page). If that all fails, install OSLEC. It's an excellent free software echo canceller, that works much better than Asterisk's default MG2. Vin?cius Fontes www.asteriskforum.com.br - Informa??es e discuss?o sobre Asterisk e telefonia IP ----- "Jason Baker" <jbaker at glastender.com> escreveu:> Well I tried Doug's suggestion and the echo is now better, but when I > call an outside analog line I still get some echo. I can hear my voice > in the ear piece of the phone with a slight delay. Is this just a fact > of life with VoIP, or is there a better way to reduce line echo? > > Again, for reference, I am using the VPMADT032 echo cancellation > module attached to a Digium TE121 PCI express card. The incoming phone > service is a PRI. > > > > Jason Baker > IT Coordinator Glastender, Inc. > 5400 North Michigan Road > Saginaw, Michigan 48604 USA > Phone: 989.752.4275 ext. 228 > Fax: 989.752.4276 > www.glastender.com > > Doug Lytle wrote: > > Jason Baker wrote: > > language = en > > group = 1 > echocancel = yes > echotraining = yes > signalling = pri_cpe > switchtype = 4ess > usecallerid = yes > context = incoming > channel => 1-23 Just noted that your system is out of Saginaw. The > system below is out > of Livonia, with an AT&T PRI as well. Note the rx/txgain entries, it > may be useful as well: > > > switchtype=national > context=pri > signalling=pri_cpe > echocancel=yes > echotraining=yes > echocancelwhenbridged=yes > rxgain=-1.0 > txgain=-4.0 > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
Oops sorry, I didn't read you have a VPMADT032 module. (Damn ADD) Don't bother setting echocancel=256, echocancel=yes should be working fine. Also don't bother with OSLEC. Maybe the IP part of the call is introducing the delay that is being perceived as echo. Do you have QoS set up to minimize latency on the IP side? Vin?cius Fontes www.asteriskforum.com.br - Informa??es e discuss?o sobre Asterisk e telefonia IP ----- "Jason Baker" <jbaker at glastender.com> escreveu:> Well I tried Doug's suggestion and the echo is now better, but when I > call an outside analog line I still get some echo. I can hear my voice > in the ear piece of the phone with a slight delay. Is this just a fact > of life with VoIP, or is there a better way to reduce line echo? > > Again, for reference, I am using the VPMADT032 echo cancellation > module attached to a Digium TE121 PCI express card. The incoming phone > service is a PRI. > > > > Jason Baker > IT Coordinator Glastender, Inc. > 5400 North Michigan Road > Saginaw, Michigan 48604 USA > Phone: 989.752.4275 ext. 228 > Fax: 989.752.4276 > www.glastender.com > > Doug Lytle wrote: > > Jason Baker wrote: > > language = en > > group = 1 > echocancel = yes > echotraining = yes > signalling = pri_cpe > switchtype = 4ess > usecallerid = yes > context = incoming > channel => 1-23 Just noted that your system is out of Saginaw. The > system below is out > of Livonia, with an AT&T PRI as well. Note the rx/txgain entries, it > may be useful as well: > > > switchtype=national > context=pri > signalling=pri_cpe > echocancel=yes > echotraining=yes > echocancelwhenbridged=yes > rxgain=-1.0 > txgain=-4.0 > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users