[A*UG] local at&t pstn/pots AC terminal impedance values

[Brett Nemeroff]

Hi Ryan,To suggest the group has been quiet for a while is quiet
an understatement. Apparently it's been so dead for so long, the mailing
list has been taken over by a Ruby group. Go figure.

Where are you seeing these options? If you are hearing the echo in your
GXP-2000, chances are you can't fix the problem. If the OTHER party hears
the echo, then you probably can. Think of it this way.. What is the echo?
It's your voice from your GXP-2000 traveling all the way to your
destination.. The someone wrong over there causes your voice to come back.
The propagation delay is perceived and thus, you hear echo. Without
propagation delay, it simply sounds like line voltage.

Now that's true of traditional telephony. With VoIP in the mix, there's a
bazillion things that can go wrong that can effect echo on either end. But
typically none of those would require changing impedance settings on your
local end.

What's your handoff to the PSTN? Can you describe your setup?  For what it's
worth, you typically don't need to mess with that kind of setting *ever*
unless you are:
1. Not in the US
2. Behind a really really really old switch
3. On a ridiculously long loop in the middle of nowhere.

Also, I don't mean to be a stick in the mud, but Grandstream has
traditionally built some of the cheapest VoIP components out there.. And I
mean cheap.. not inexpensive. You may want to try a higher quality device to
see if you are happier. That being said, I *know* the GXP-2000 is better
than many of their other devices and I do NOT have any experience with it.

That's my $0.02. Feel free to correct me if I'm wrong. :)

BTW, Is there any community interest in reviving this group? Personally, I'm
not involved in Asterisk as much as I used to be because I'm trying to focus
on more core solutions and using OpenSER/OpenSIPs/Kamailio/Whatever you want
to call it these days. I'm curious what everyone else is doing with VoIP in
general out there and how we may all be able to benefit from our collective
experiences.

I for one, am looking for some people who really know what they are doing
with Asterisk, and SIP in general, to provide some redundant support to my
clients.
-Brett




On Tue, Sep 30, 2008 at 4:07 PM, Ryan Vanderwerf <
rvanderwerf at kpi-consulting.net> wrote:

> Hi, new to the group. I've set up a recent system, and I'm getting a lot of
> echo on a GXP-2000 phone through a PSTN line at the value of 600R- 600 ohms.
> My options are:
> 600R - 600 ohms</option>
> 600C - 600 ohms + 2.16uF</option>
> 900R - 900 ohms</option>
> 900C - 900 ohms + 2.16uF</option>
> COMPLEX1 - 220 ohms + (820 ohms || 115nF)
> COMPLEX2 - 270 ohms + (750 ohms || 150nF)
> COMPLEX3 - 370 ohms + (620 ohms || 310nF)
> COMPLEX4 - 600R 270 ohms + (750 ohms || 150nF)
> COMPLEX5 - 320 ohms + (1050 ohms || 230nF)
> COMPLEX6 - 350 ohms + (1000 ohms || 210nF)
> COMPLEX7 - 200 ohms + (680 ohms || 100nF)
> COMPLEX8 - 370 ohms + (820 ohms || 110nF)
> COMPLEX9 - 275 ohms + (780 ohms || 115nF)
> COMPLEX10 - 120 ohms + (820 ohms || 110nF)
>
> I'm trying to play the the AC terminal impedance values, and I was
> wondering if anyone knew what value the local Austin area AT&T PSTN/Pots
> lines use? I figured I'd ask before trying all of them.
>
> Thanks
>
> Ryan Vanderwerf
> _______________________________________________
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> Austin-Asterisk-Users-Group at bybent.com
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>
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