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.



Vinícius Fontes
www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP

----- "Jason Baker" <[email protected]> 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 
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