Re: T1 'access layer' used Cisco or new Digium

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T1 'access layer' used Cisco or new Digium
\ Tom Browning (16 Feb 2008)
. \ Steve Edwards (16 Feb 2008)
. . \ Steve Totaro (16 Feb 2008)
. . . \ Steve Edwards (16 Feb 2008)
. \ Chris Bagnall (16 Feb 2008)

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Subject:	Re: T1 'access layer' used Cisco or new Digium
Group:	Asterisk-users
From:	Steve Edwards
Date:	16 Feb 2008

On Sat, 16 Feb 2008, Steve Totaro wrote:

> On Feb 16, 2008 1:14 PM, Steve Edwards <asterisk.org> wrote:
>> On Sat, 16 Feb 2008, Tom Browning wrote:
>>
>>> b) buy Digium T1 cards in 2 port or 4 port flavors and place in 1U or 2U
>>> rackmount servers and use as dedicated ISDN T1 to SIP gateways.
>>
>> I did this for a client a couple of years ago. te410p's in 1u's
>> (Supermicro at the time HP DL380's now). I ran IAX instead of SIP. The
>> "telco servers" answered the calls and dialed to an "application server"
>> that did all the voice processing.
>>
>> Client is still happy.
>>
>> Thanks in advance,
>> ------------------------------------------------------------------------
>> Steve Edwards sedwards Voice: +1-760-468-3867 PST
>> Newline Fax: +1-760-731-3000
>>
>
> I did the same thing with a T3 terminated into an Adtran MX2800 M13
> that broke off into 28 T1s that terminated into HP DL320s with quad
> port Sangoma boards (no echo can) and handed the calls off as SIP to
> the application server.

Do all 672 calls go to a single server? I've never had more than 200 calls
to a single server and it seemed pretty busy :)

Why did you choose SIP over IAX? I chose IAX because it was easier to
configure. Also I thought IAX would be less "chatty" and "trunking" would
reduce network traffic -- even though the servers are all in the same
rack.

Thanks in advance,
------------------------------------------------------------------------
Steve Edwards sedwards Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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