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<DIV dir=ltr align=left><SPAN class=549044906-10062007><FONT face=Arial
size=2>Best way to do this is not touch the sip.cfg, ever. Leave it as
included in each release and include your overrides in a different
file.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=549044906-10062007><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=549044906-10062007><FONT face=Arial
size=2>Then reference your files like this in your MAC.cfg file, your file will
override the sip.cfg defaults.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=549044906-10062007><FONT face=Arial
size=2>CONFIG_FILES="phone_user.cfg,server.cfg,sip.cfg"</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=549044906-10062007><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=549044906-10062007><FONT face=Arial
size=2>In server.cfg, if you wanted to change the server, for
example:</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=549044906-10062007><FONT face=Arial
size=2><?xml version="1.0"
standalone="yes"?><BR><sip><BR>
<voIpProt><BR> <local
voIpProt.local.port=""/><BR> <server
voIpProt.server.1.address="asterisk.yourdomain.com" <BR>
</voIpProt><BR></sip></FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=549044906-10062007></SPAN><SPAN
class=549044906-10062007><FONT face=Arial size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=549044906-10062007></SPAN> </DIV>
<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left>
<HR tabIndex=-1>
<FONT face=Tahoma size=2><B>From:</B> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <B>On Behalf Of </B>C
F<BR><B>Sent:</B> Saturday, June 09, 2007 22:54<BR><B>To:</B> Asterisk Users
Mailing List - Non-Commercial Discussion<BR><B>Subject:</B> Re: [asterisk-users]
Bad Echo between SIP calls<BR></FONT><BR></DIV>
<DIV></DIV>It doesn't matter if it's supported, they are all, however I have
seen some echo problems after firmware upgrades, the only way to fix it was to
either copy the differences or overwrite my old config files with the new ones
that came with the firmware and then modify as needed for my setup.<BR><BR>
<DIV><SPAN class=gmail_quote>On 6/10/07, <B class=gmail_sendername>Deepak
Naidu</B> <<A href="mailto:deepak_nai@yahoo.com">deepak_nai@yahoo.com</A>>
wrote:</SPAN>
<BLOCKQUOTE class=gmail_quote
style="PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid">
<DIV>The sip config & firmware are the supported one for the existing
firmware. If you have any stable working Polycom 501 SIP without echo
between SIP-->SIP & wouldnt mind to share the sip.cfg, sip.ld
& bootrom would be great, bcos I have not got concreate resolution for
this issue.</DIV>
<DIV> </DIV>
<DIV>Hope I can resolve this mess. Feels bad when one does best in
aggregating things & some louzy device screws up... Oh my frustation is
comming on mail :<IMG
src="http://us.i1.yimg.com/us.yimg.com/i/mesg/tsmileys2/03.gif"
NOSEND="1"></DIV>
<DIV> </DIV>
<DIV> </DIV>
<DIV>--</DIV>
<DIV>Deepak<SPAN class=q><BR><BR><B><I>C F <<A
onclick="return top.js.OpenExtLink(window,event,this)"
href="mailto:shmaltz@gmail.com" target=_blank>
shmaltz@gmail.com</A>></I></B> wrote:</SPAN></DIV>
<BLOCKQUOTE
style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: rgb(16,16,255) 2px solid"><SPAN
class=q>Are the config files you are using with the phones what was meant
with <BR>that firmware? or did you upgrade the firmware and reused the
old<BR>config files?<BR><BR></SPAN><SPAN class=q>On 6/9/07, Steve Underwood
wrote:<BR>> Stephen Davies wrote:<BR></SPAN>
<DIV><SPAN class=e id=q_1131403358693ff4_5>> > On 09/06/07, Deepak
Naidu wrote:<BR>> >> Ya, I have done that, below is zapata.conf .
Also we had an TMP card<BR>> >> with<BR>> >> analog lines.
& SIP cals were great on them. & now when we switched<BR>>
>> 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>> By
this measure most phones are nasty. The handset should be echo<BR>>
cancelled, to prevent leakage of the earpiece into the mike. It is <BR>>
getting less and less common to do this, now. Polycoms, Sipuras,
Snoms,<BR>> you name it, they do it badly. Many are not too annoying
until someone<BR>> turns the volume up. Call someone a little hard of
hearing and you will <BR>> hear echo.<BR>><BR>>
Steve<BR>><BR>><BR>>
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