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 <[EMAIL PROTECTED]> 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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 <[EMAIL PROTECTED]> 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
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
   
    
    
---------------------------------
  
  Yahoo! Answers - Get better answers from someone who knows. Try it now.


_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users


       
---------------------------------
 Yahoo! Answers - Get better answers from someone who knows. Tryit now.
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to