[Asterisk-Users] Generic X100P's
Steve Edwards
asterisk.org at sedwards.com
Mon Oct 11 17:49:52 MST 2004
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
> Asterisk-Users at lists.digium.com
> 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 sedwards at sedwards.com Voice: +1-760-468-3867 PST
Newline pagesteve at sedwards.com Fax: +1-760-731-3000
More information about the asterisk-users
mailing list