[Asterisk-Users] PSTN > SIP Gateways?

[Jess Magnaye]

i worked with Mediatrix boxes before (in '01-'02).  configuring them using
the ume (unit manager express) is highly suggested, as there are
"parameters" in there that you won't see in its telnet access... :)

generally it works fine, depending on how you use them, or how strict are
you in evaluating a product.  i used it for both origination (fxs) and
termination (fxo). for termination, they only supported battery-reversal
that time. i am not aware now if they support answer supervision via
voice-detect. a particular problem i remember with the termination is that
it cannot handle properly the "far-end disconnect", where it locks up or the
channel appears to be in a call even if the far-end already got
disconnected. for fxs, i had problems with the volume (rx/tx gain) -- 
they're inconsistent -- something like, you'll find them working at its best
today, and the next day you don't know what happens why the quality is not
the same as it was.

(so maybe they made some improvements at the present?.... u have to find
out. :) good luck. )


----- Original Message ----- 
From: "Michael Graves" <mgraves at mstvp.com>
To: <asterisk-users at lists.digium.com>
Sent: Friday, January 09, 2004 5:56 PM
Subject: [Asterisk-Users] PSTN > SIP Gateways?


> Since my earlier inquiry about gateways went unanswered perhaps
> rephrasing will help. Does anyone here have experience with standalone
> SIP FXO gateways like those from Mediatrix? Care to share their
> experiences with them? Off list if necessary.
>
> Michael
>
> --
> Michael Graves                           mgraves at pixelpower.com
> Sr. Product Specialist                          www.pixelpower.com
> Pixel Power Inc.                                 mgraves at mstvp.com
>
> "I knew a man who couldn't lose 'cause he never gave in. He stuck
> to his pistols and it made him a better man." - D. Van Zant
>
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