[Asterisk-Users] SIP and ECHO

[Dave Alan Caruana]

hi ..
i have the exact same problem you have ..
seems to be related to Budgettone phones in my prob.
I *tried* selling an asterisk exchange to a client
and today he phoned telling me he is very unsatisfied
& I risk being thrown out .. suggestions would be 
welcome! i've tried *everything* that has gone 
in your correspondence on the list, and a few
of my own .. no luck! seems a hardware problem.

arghhh!!!

the other budgettone problem is it won't do a 
consultative transfer ie. you answer an incoming call,
speak to someone else on a different extension 
and then pass the call .. only way I have found to
"emulate" that is using call parking which is VERY
messy!!

well .. 
maybe it's consoling to know you are not alone!

cheers,
Dave

  ----- Original Message ----- 
  From: Daniel ANDRE 
  To: asterisk-users at lists.digium.com 
  Sent: Friday, August 29, 2003 5:56 PM
  Subject: Re: [Asterisk-Users] SIP and ECHO


  Hello,

  Brian West a écrit:

I get no echo on mine.. but you can check to make sure your line isn't
reversed.  A reverse wired jack can do that.
  I don't think so but I have tested reversed and it doesn't solve my echo problem

  Daniel


bkw

On Thu, 28 Aug 2003, Brian J. Schrock wrote:

  
I can minimize doing those tricks, but I cannot seem to get it to go
away.

On Thu, 2003-08-28 at 11:33, Dan wrote:
    
----- Original Message -----
From: "Brian J. Schrock" <brians at anistonetech.com>
To: <asterisk-users at lists.digium.com>
Sent: Thursday, August 28, 2003 6:16 PM
Subject: [Asterisk-Users] SIP and ECHO


      
Hello,

I have read the information on echo and SIP in the FAQ and I have
scoured the mailing list for possible solutions, but as yet I have not
been able to get rid of this echo.

I have a Sip phone, Digium 4 Port FXS, and 3 Digium FXO cards installed
into an asterisk server. If I call between the Sip Phone
(Budgettone-100) and the 4 FXS ports everything sounds great. If I call
out to the PSTN through the FXO cards I get horrible echo, I have even
been able when talking loud enough to get a horrible feedback loop
going. I have tried 4 different echo cancellers in the Makefile for the
Zap drivers and nonoe of them changed the situation.

I have echocancel = (Any where from 1 - 256, I have tried alot of
different values), and I have echocanelwhenbridged = yes.I only hear the
echo start when the call gets bridged onto the outgoing PSTN lines.

Is there anything I can do?

Brian J. Schrock

        
Hi,

For me:

rxgain=0.8
txgain=0.8

in zapata conf do the trick.
Now the echo is allmost inexistant. Maybe the sound is not very strong but
the quality is very good.
I have the default echo canceller (no modification in the source files).

Tested with a lot of SIP phones (ATA (G.711), X-Lite(GSM), SJ_phone(G.711),
Cisco 79x0) and one X100P card.

BR,
Dan

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-- 
Daniel ANDRE (mailto:dandre at iris-tech.fr)
IRIS Technologies - http://www.iris-tech.com
Serveur kwartz - http://www.kwartz.com


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