[asterisk-users] Bad Echo between SIP calls

[Steve Totaro]

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> bounces at lists.digium.com] On Behalf Of Stephen Davies
> Sent: Saturday, June 09, 2007 4:00 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Bad Echo between SIP calls
> 
> On 09/06/07, Deepak Naidu <deepak_nai at yahoo.com> 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


Most likely the phones.  Is it worse on speakerphone?  Are they cheap
like the Grandstream 101s?  Try with a couple softphones and headsets,
any better.

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB