Need recommendation for an analog interface card

I have just 1 POTS line in a small office environment.
I don’t need more POTS lines.
Internally, we have 4 extensions and that is enough.
This setup is not going to change because we mainly
communicate via email and text chat on our DSL broadband connection.

The DSL and analog phone run on the same physical line with filters.

We have had various noise and echo problems with the x100P card.
I am thinking about dumping the x100p card.

Give the above setup - Can you recommend the next best analog interface card?

Thanks.

clone x100 cards are frequently to blame for a wide variety of issues.

You would want a Sangoma A200 series (no echo canceller) or Digium TDM400 series, one FXO port. (you’d have to get two ports on the sangoma, they spec out by twos). Both shoudl work well for you…

Thanks. I am looking into the Sangoma and Digium cards.

Currently, the x100p is connected to the POTS like this:
POTS Wall RJ11 port <–> x100p card

In the meantime, I was wondering are there any external Echo Cancellation devices that you can utilize like this:
POTS Wall RJ11 port <–> Echo Cancellation device <–> x100p card
So that the echo is minimized before it even gets to the x100p card.

Before you go buying new hardware, you might want to pinpoint where the echo is occurring. You mention that you have 4 extensions. These are network based? You also mention that you use the network a lot. What type is it - 10BaseT, 100BaseT? What kind of switches? Might the problem be on the network? Is the echo constant or intermittent? If no one is on the network do you still have echo problems?

If you have ruled out the network, have you tried adjusting the echo cancellation on the Asterisk server?

network wouldn’t cause echo… network would cause jitter, delays, and chop, but not echo.

really really cheap IP phone on the other end can cause echo, especially with badly made speakerphones

If you are using FXS ports and phones, that could also be where the echo is.

A few things to try also-

  1. run fxotune. (check voip-info.org for details)
  2. in zapata.conf turn on echo cancel, echocancelwhenbridged and echotraining.
  3. in zaptel source, change to mark2 echo canceller and enable aggressive echo cancellation (check voip-info for instructions)…

You can try this now with your existing card to see if it helps…

There is no ‘echo cancellation device’ you can wire between the line and the card, this would be most inefficient and the card can generally do a better job. It would also make configuration much more troublesome.

We have a 100BaseT network.
This is on one 8 port switch behind the firewall.

The 4 extensions are a combination of
2 ATAs (with traditional analog phones) and
2 X-Lite SIP Softphones

We have no IP phones.

We get the echo and noise on all 4 extensions.
You can physically disconnect any node(s) from the network and
still get the echo and noise as long as any remaining extension is running.

We swapped the 8 port switch with another one. It made no difference.

Yes, we experimented with fxotune, zapata.conf and zaptel source
(including fixing the spinlock.h file, make install, etc.)
We noticed some very minor differences but nothing worth the trouble.

The ONLY thing that made the difference was:

  1. On a second identical PC we reinstalled the whole system.
  2. But this time without the x100p card.
  3. Just added and configured the same 4 extensions.
  4. We still used the same network, firewall, switch, etc.

Since there was no connectivity to the POTS line all
the call activity was between the 4 extensions.
There was NO echo or noise.

Then we installed the x100p card into the 2nd PC and
the echo and noise started all over again.

We have replaced the x100p card 2 times already. Same problems.

We even shutdown and disconnected the DSL modem from the wall.
With or without the filters, we still have the echo and noise
problems as long as the x100p is involved.

Then we called our DSL service provider and asked them to run their
tests on our line. They ran their tests with our DSL modem and
x100p connected, individually connected and both disconnected.
They said the results were within their “normal” range. Nothing unusual.

We think the x100p is the culprit.

So we are reviewing the Sangoma and Digium cards as recommended by IronHelix.

quick question-

if the x100 card is to blame, then you should in any configuration get NO echo or very little echo between internal extensions (between ata/softphone/etc). The echo should only appear when using the outbound line, correct?

try the aggressive echo cancel thing. you must edit a line in the zaptel source, then make/make install the zaptel package. It helps sometimes…

Otherwise yeah give the TDM400 or A200 a shot. Good luck!

[quote=“IronHelix”]quick question-

if the x100 card is to blame, then you should in any configuration get NO echo or very little echo between internal extensions (between ata/softphone/etc). The echo should only appear when using the outbound line, correct?[/quote]Yes, this is correct.

[quote]try the aggressive echo cancel thing. you must edit a line in the zaptel source, then make/make install the zaptel package. It helps sometimes…

Otherwise yeah give the TDM400 or A200 a shot. Good luck![/quote]I assume you are suggesting that I do
#define ECHO_CAN_MARK2
with
#define AGGRESSIVE_SUPPRESSOR

Correct?

not quite. in zconfig.h, you will see many echo cancellers defined. some are commented out with /* commented out stuff /. The compiler ignores everything between / and */.

You want to remove the comments around aggressive_suppressor, so it is like this:

/*
 * Uncomment for aggressive residual echo supression under
 * MARK2 echo canceller
 */
#define AGGRESSIVE_SUPPRESSOR

then uncomment the
#define ECHO_CAN_MARK2
making sure all the other echo cancellers have comments (/* */) around them.

I tried the aggressive suppressor.

The dmesg confirms it:
Zapata Telephony Interface Registered on major 196
Zaptel Version: 1.2.7 Echo Canceller: MARK2 (aggressive)

It did not help.
With the x100p, any PSTN call has echo and noise.

But the calls between extensions are clear.

To order a TDM400 or A200, we need to get internal authorization on a weekday.
It is still Sunday here.

While we wait - could you please give your opinion on the following -

[color=brown]The Asterisk system has an IP address of 192.168.5.3
It has been working great so far. (Excluding the x100p problems.)

Lets assume that we removed the x100p card and
reinstall and reconfigured Asterisk with the 4 extensions.

(Remember, we have 2 ATAs with traditional analog phones.)

We want to connect the POTS to Asterisk, in order to:
a. Allow any of the 4 extensions to make ougoing local calls via Asterisk to PSTN.
b. Allow incoming local PSTN calls to be routed to the 4 extensions via Asterisk.

How about using the Grandstream HT488 to enable connectivity
between Asterisk and the POTS line.
See grandstream.com/y-ht488.htm

I know the HT488 is an ATA but according to the HT488 User Manual:
See grandstream.com/user_manuals … Manual.pdf

  1. It has a FXO RJ-11 port to connect to PSTN

  2. It has a LAN RJ-45 port (which could be assigned an IP address of 192.168.5.4)

  3. On page 14 of the User Manual, “5.2.9 VoIP-to-PSTN Calls. This function is
    applicable on FXO port that functions as a bridge between VoIP and PSTN. The
    user can remotely use PSTN line to initiate a call.”

  4. On page 14 of the User Manual, “5.2.10 PSTN-to-VoIP Calls. This function is
    applicable on FXO port that functions as a bridge between VoIP and PSTN. The
    user can make VoIP calls remotely by dialing into FXO Line port on HT488.”

So is it possible to use the Grandstream HT488 to do the above?
Instead of installing a PCI Analog Interface card in the Asterisk PC?

This HT488 is much less expensive than the Sangoma or Digium PCI cards.
(We will not need authorization to procure it.)
It comes with echo and noise cancellation within the hardware.

We are already using the Grandstream HT286 ATA, it has no FXO port but it works great as a basic ATA.[/color]

yes this is possible. Also consider the Sipura/Linksys SPA3000 type ATA, it does the same thing…

Just keep in mind that it adds an extra level of complexity (you have to tweak the ATA to work nicely on your line, and you also have to set up SIP to connect to the * server…)

also watch your dialplan, sipura dialplans are tricky but you can do some cool stuff with them…

Thanks IronHelix.

I am looking at the Sipura/Linksys SPA3000.