[asterisk-users] Bad Echo between SIP calls

[Steve Totaro]

Do you have reinvites enabled?  Are you running this over a linksys four
port SoHo router/switch or something?

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
<http://www.asteriskhelpdesk.com/component/option,com_wrapper/Itemid,37/
> 
KB3OPB
  

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From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Deepak
Naidu
Sent: Saturday, June 09, 2007 4:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bad Echo between SIP calls

 

Steve I understand your theory.  We have Poycom 501 phones.  Prior
upgrading to PRI we were till date using 4 analog lines connected with
TDM card from digium & no echo for pure SIP to SIP lines.

 

Now I have TE212P which had onboard echo cancellor.

 

I am trying make myself clear before I blame on any network.  B'cos for
sure we have a spegati of networks & no QoS.  Also the intresting thing
is if I call from one extension to other dialing the main line & then
extension the call is crystal clear.  but when dialing a direct
extension its a hell of echo.

 

--

Deepak

Stephen Davies <stephen.l.davies at gmail.com> wrote:

	On 09/06/07, Deepak Naidu wrote:
	> 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.
	
	If you are getting echo on pure SIP to SIP calls, there's no
point in
	fiddling around with your zapta.conf. That file is for
configuring
	chan_zap, which is used to talk to Zap/ channels. Your calls are
SIP
	to SIP so the zap channel and your PRI aren't being used at all.
	
	SIP calls are "pure digital" 4 wire lines so no electrical
(Hybrid)
	echo will be present. The phones should not generate echo. If
they
	are, they are presumably nasty phones (what kind are they?) and
you
	should get properly made phones.
	
	Steve
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