<div>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.</div> <div> </div> <div><STRONG>canreinvite=no</STRONG></div> <div><BR>--</div> <div>Deepak</div> <div> </div> <div><BR><B><I>Steve Totaro <stotaro@asteriskhelpdesk.com></I></B> wrote:</div> <BLOCKQUOTE class=replbq style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid"> <META content="Microsoft Word 11 (filtered medium)" name=Generator> <STYLE> v\:* {behavior:url(#default#VML);} o\:* {behavior:url(#default#VML);} w\:* {behavior:url(#default#VML);} .shape {behavior:url(#default#VML);} </STYLE> <?xml:namespace prefix = o ns = "urn:schemas-microsoft-com:office:office" /><o:SmartTagType name="PersonName" namespaceuri="urn:schemas-microsoft-com:office:smarttags"></o:SmartTagType> <STYLE>
st1\:*{behavior:url(#default#ieooui) } </STYLE> <STYLE> <!-- /* Font Definitions */ @font-face {font-family:Tahoma; panose-1:2 11 6 4 3 5 4 4 2 4;} /* Style Definitions */ p.MsoNormal, li.MsoNormal, div.MsoNormal {margin:0in; margin-bottom:.0001pt; font-size:12.0pt; font-family:"Times New Roman";} a:link, span.MsoHyperlink {color:blue; text-decoration:underline;} a:visited, span.MsoHyperlinkFollowed {color:blue; text-decoration:underline;} p {mso-margin-top-alt:auto; margin-right:0in; mso-margin-bottom-alt:auto; margin-left:0in; font-size:12.0pt; font-family:"Times New Roman";} span.EmailStyle18 {mso-style-type:personal-reply; font-family:Arial; color:navy;} @page Section1 {size:8.5in 11.0in; margin:1.0in 1.25in 1.0in 1.25in;} div.Section1 {page:Section1;} --> </STYLE> <DIV class=Section1> <div class=MsoNormal><FONT face=Arial color=navy size=2><SPAN style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY:
Arial">Do you have reinvites enabled? Are you running this over a linksys four port SoHo router/switch or something?<o:p></o:p></SPAN></FONT></div> <DIV> <div><FONT face="Times New Roman" color=navy size=2><SPAN style="FONT-SIZE: 10pt; COLOR: navy">Thanks,<BR>Steve Totaro<BR><A href="http://www.asteriskhelpdesk.com/component/option,com_wrapper/Itemid,37/">http://www.asteriskhelpdesk.com</A><BR>KB3OPB<BR> </SPAN></FONT><FONT color=navy><SPAN style="COLOR: navy"> </SPAN></FONT><o:p></o:p></div></DIV> <DIV style="BORDER-RIGHT: medium none; PADDING-RIGHT: 0in; BORDER-TOP: medium none; PADDING-LEFT: 4pt; PADDING-BOTTOM: 0in; BORDER-LEFT: blue 1.5pt solid; PADDING-TOP: 0in; BORDER-BOTTOM: medium none"> <DIV> <DIV class=MsoNormal style="TEXT-ALIGN: center" align=center><FONT face="Times New Roman" size=3><SPAN style="FONT-SIZE: 12pt"> <HR tabIndex=-1 align=center width="100%" SIZE=2> </SPAN></FONT></DIV> <div class=MsoNormal><B><FONT face=Tahoma size=2><SPAN
style="FONT-WEIGHT: bold; FONT-SIZE: 10pt; FONT-FAMILY: Tahoma">From:</SPAN></FONT></B><FONT face=Tahoma size=2><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma"> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <B><SPAN style="FONT-WEIGHT: bold">On Behalf Of </SPAN></B>Deepak Naidu<BR><B><SPAN style="FONT-WEIGHT: bold">Sent:</SPAN></B> Saturday, June 09, 2007 4:54 AM<BR><B><SPAN style="FONT-WEIGHT: bold">To:</SPAN></B> <?xml:namespace prefix = st1 ns = "urn:schemas-microsoft-com:office:smarttags" /><st1:PersonName w:st="on">Asterisk Users Mailing List - Non-Commercial Discussion</st1:PersonName><BR><B><SPAN style="FONT-WEIGHT: bold">Subject:</SPAN></B> Re: [asterisk-users] Bad Echo between SIP calls</SPAN></FONT><o:p></o:p></div></DIV> <div class=MsoNormal><FONT face="Times New Roman" size=3><SPAN style="FONT-SIZE: 12pt"><o:p> </o:p></SPAN></FONT></div> <DIV> <div class=MsoNormal><FONT face="Times New Roman" size=3><SPAN
style="FONT-SIZE: 12pt">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.<o:p></o:p></SPAN></FONT></div></DIV> <DIV> <div class=MsoNormal><FONT face="Times New Roman" size=3><SPAN style="FONT-SIZE: 12pt"> <o:p></o:p></SPAN></FONT></div></DIV> <DIV> <div class=MsoNormal><FONT face="Times New Roman" size=3><SPAN style="FONT-SIZE: 12pt">Now I have TE212P which had onboard echo cancellor.<o:p></o:p></SPAN></FONT></div></DIV> <DIV> <div class=MsoNormal><FONT face="Times New Roman" size=3><SPAN style="FONT-SIZE: 12pt"> <o:p></o:p></SPAN></FONT></div></DIV> <DIV> <div class=MsoNormal><FONT face="Times New Roman" size=3><SPAN style="FONT-SIZE: 12pt">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.<o:p></o:p></SPAN></FONT></div></DIV> <DIV> <div class=MsoNormal><FONT face="Times New Roman" size=3><SPAN style="FONT-SIZE: 12pt"> <o:p></o:p></SPAN></FONT></div></DIV> <DIV> <div class=MsoNormal><FONT face="Times New Roman" size=3><SPAN style="FONT-SIZE: 12pt">--<o:p></o:p></SPAN></FONT></div></DIV> <DIV> <div class=MsoNormal><FONT face="Times New Roman" size=3><SPAN style="FONT-SIZE: 12pt">Deepak<BR><BR><B><I><SPAN style="FONT-WEIGHT: bold; FONT-STYLE: italic">Stephen Davies <stephen.l.davies@gmail.com></SPAN></I></B> wrote:<o:p></o:p></SPAN></FONT></div></DIV> <BLOCKQUOTE style="BORDER-RIGHT: medium none; PADDING-RIGHT: 0in; BORDER-TOP: medium none; MARGIN-TOP: 5pt; PADDING-LEFT: 4pt; MARGIN-BOTTOM: 5pt; PADDING-BOTTOM: 0in; MARGIN-LEFT: 3.75pt; BORDER-LEFT: #1010ff 1.5pt solid;
PADDING-TOP: 0in; BORDER-BOTTOM: medium none"> <div class=MsoNormal><FONT face="Times New Roman" size=3><SPAN style="FONT-SIZE: 12pt">On 09/06/07, Deepak Naidu <DEEPAK_NAI@YAHOO.COM>wrote:<BR>> Ya, I have done that, below is zapata.conf. Also we had an TMP card with<BR>> analog lines. & SIP cals were great on them. & now when we switched over.<BR>> SIP calls have echo.. which shouldnt be at all.<BR><BR>If you are getting echo on pure SIP to SIP calls, there's no point in<BR>fiddling around with your zapta.conf. That file is for configuring<BR>chan_zap, which is used to talk to Zap/ channels. Your calls are SIP<BR>to SIP so the zap channel and your PRI aren't being used at all.<BR><BR>SIP calls are "pure digital" 4 wire lines so no electrical (Hybrid)<BR>echo will be present. The phones should not generate echo. If they<BR>are, they are presumably nasty phones (what kind are they?) and you<BR>should get properly made
phones.<BR><BR>Steve<BR>_______________________________________________<BR>--Bandwidth and Colocation provided by Easynews.com --<BR><BR>asterisk-users mailing list<BR>To UNSUBSCRIBE or update options visit:<BR>http://lists.digium.com/mailman/listinfo/asterisk-users<o:p></o:p></SPAN></FONT></div></BLOCKQUOTE> <div class=MsoNormal><FONT face="Times New Roman" size=3><SPAN style="FONT-SIZE: 12pt"><o:p> </o:p></SPAN></FONT></div> <div><FONT face="Times New Roman" size=3><SPAN style="FONT-SIZE: 12pt"> <o:p></o:p></SPAN></FONT></div> <DIV class=MsoNormal style="TEXT-ALIGN: center" align=center><FONT face="Times New Roman" size=3><SPAN style="FONT-SIZE: 12pt"> <HR align=center width="100%" SIZE=1> </SPAN></FONT></DIV> <div class=MsoNormal><FONT face="Times New Roman" size=3><SPAN style="FONT-SIZE: 12pt">Yahoo! Answers - Get better answers from someone who knows. <A
href="http://uk.answers.yahoo.com/;_ylc=X3oDMTEydmViNG02BF9TAzIxMTQ3MTcxOTAEc2VjA21haWwEc2xrA3RhZ2xpbmU">Try it now</A>.<o:p></o:p></SPAN></FONT></div></DIV></DIV>_______________________________________________<BR>--Bandwidth and Colocation provided by Easynews.com --<BR><BR>asterisk-users mailing list<BR>To UNSUBSCRIBE or update options visit:<BR>http://lists.digium.com/mailman/listinfo/asterisk-users<BR></BLOCKQUOTE><BR><p> 
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