[asterisk-users] Bad Echo between SIP calls

[Darryl Dunkin]

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.
 
Then reference your files like this in your MAC.cfg file, your file will
override the sip.cfg defaults.
CONFIG_FILES="phone_user.cfg,server.cfg,sip.cfg"
 
In server.cfg, if you wanted to change the server, for example:
<?xml version="1.0" standalone="yes"?>
<sip>
   <voIpProt>
      <local voIpProt.local.port=""/>
      <server voIpProt.server.1.address="asterisk.yourdomain.com" 
   </voIpProt>
</sip>
 
 
________________________________

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of C F
Sent: Saturday, June 09, 2007 22:54
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bad Echo between SIP calls


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.


On 6/10/07, Deepak Naidu <deepak_nai at yahoo.com> wrote: 

	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.
	 
	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 :
<http://us.i1.yimg.com/us.yimg.com/i/mesg/tsmileys2/03.gif> 
	 
	 
	--
	Deepak
	
	C F < shmaltz at gmail.com <mailto:shmaltz at gmail.com> > wrote:

		Are the config files you are using with the phones what
was meant with 
		that firmware? or did you upgrade the firmware and
reused the old
		config files?
		
		On 6/9/07, Steve Underwood wrote:
		> Stephen Davies 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.
		> By this measure most phones are nasty. The handset
should be echo
		> cancelled, to prevent leakage of the earpiece into the
mike. It is 
		> getting less and less common to do this, now.
Polycoms, Sipuras, Snoms,
		> you name it, they do it badly. Many are not too
annoying until someone
		> turns the volume up. Call someone a little hard of
hearing and you will 
		> hear echo.
		>
		> Steve
		>
		>
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