Steve -

When you recommend a channel bank to eliminate echo, are you assuming use of a T-1 with the channel bank? I'm curious because I bought a used CAC ADIT600 with 24 FXO lines and 8 FXS. I'm terminating my 8 POTs lines into 8 of the FXO lines on the ADIT600 as I don't have a T-1. The ADIT600 is connected to my Asterisk box via a T100P. I'm using Cisco 7960 IP phones. I'm having local echo problems (customer calling in can't hear the echo but my agents hear it). Thus, I'm using a channel bank and I've got the problem.

--On Monday, October 11, 2004 5:49 PM -0700 Steve Edwards <[EMAIL PROTECTED]> wrote:

On Mon, 11 Oct 2004, Erik Espinoza wrote:

Just want to test with real phone lines to ensure that they
work with our existing pbx before deploying with 8 lines using the
tdm400p's

Consider using a t100p and a channel bank. Here's the breakdown:

        TDM                             T1
        --------------------            ----------------------------
        2 x tdm40b      $610            1 x t100p               $495
                                        1 x channel bank*       $200
                                        1 x breakout box        $100
        --------------------            ----------------------------
                        $610                                    $795

* I purchased a used (looks new to me) Adtran TA750 with 4 fxo and 20 fxs
ports for $225 a couple of weeks ago. I've seen loaded TA750's go for
between $150 and $350. An Adtran 600 would also be a good choice.

For the additional $185, you get 16 additional ports, no echo problems,
and no interupt problems.

I chose a breakout box for my on-the-road demo kit. You may already have
a 66 block or a 110 block wired up.

Also, if you can get the fxo lines off your old pbx on a t1, you may
shave off a bit more.

I've got 1 host with a tdm40b and an x101p and another host with the
t100p and the channel bank. The "T1" host is rock solid.

On Mon, 11 Oct 2004, Erik Espinoza wrote:

If reply all actually responds to the reply-to header and reply doesn't,
your MUA is broken.

There is no reply-to header being added in from the asterisk-users mailing list, I double checked this by looking through other peoples posts. The MUA works fine with mailing lists that actually add the reply-to header.

2 cards is the highest number recommended. But as I mentioned, it won't
be completely representative of your suggested final deployment and may
cause you unforeseen trouble. If you are being serious about testing for
real deployments, you should go ahead and buy final hardware. If you are
testing to deploy for a customer, you need to be very aware of your
final hardware.

Thanks for the heads up, I'll keep that in mind. I'm building a system that is just going to recieve user calls for help and call the person who is supposed to be on call for the night. This will be tested amongst our managers for a few days, making sure that calls ar routed properly and that the software works as promised before purchasing the final hardware. The agi's are all written and in the 'all voip' system work fine. Just want to test with real phone lines to ensure that they work with our existing pbx before deploying with 8 lines using the tdm400p's _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users


Thanks in advance, ------------------------------------------------------------------------ Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline [EMAIL PROTECTED] Fax: +1-760-731-3000 _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users



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