[asterisk-users] Bad Echo between SIP calls

[Deepak Naidu]

reinvite is disabled.  Also its a Dell PowerEdge 850 server running asterisk connected to a Cisco switch.  & other network in company have Cisco Switch.  Also we have approx 75 Polycoms all over.
   
  canreinvite=no
  
--
  Deepak
   
  
Steve Totaro <stotaro at asteriskhelpdesk.com> wrote:
        v\:* {behavior:url(#default#VML);}  o\:* {behavior:url(#default#VML);}  w\:* {behavior:url(#default#VML);}  .shape {behavior:url(#default#VML);}        st1\:*{behavior:url(#default#ieooui) }                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
KB3OPB
  

        
---------------------------------
  
  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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