[Asterisk-Users] Planning Asterisk

[Jay Milk]

Thanks for the complete and detailed answer.  Now gathering hardware so
I can get going.  Hope I'll get this done before baby comes :)

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Howard White
Sent: Sunday, April 25, 2004 7:16 PM
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Planning Asterisk


Jay,
Your long awaited response to your planning questions :)


On Friday 23 April 2004 16:59, Jay Milk wrote:
> Hello,
>
> I'm planning to convert my phone system to Asterisk, as I've outgrown 
> my TalkSwitch system.  I have a few questions for experienced * users,

> most of which can be answered yes/no.
>
> Current Setup:
> - Talkswitch 48NLS (4CO/8Ext) phone system.
> - One CO line, two Vonage lines, one Voicepulse line connected to 
> phone system
> - A third Vonage line directly connected to a fax machine
> - A sipgate.de line connected through Port#2 of VoicePulse's Sipura to

> a stand-alone phone.
>
> Getting VoicePulse (recently) and finding sipgate.de pushed me over 
> the 4-line limit of the Talkswitch PBX, plus there are some 
> shortcomings to Talkswitch which I could, but don't want to live with.
>
> To get my CO and SIP lines connected I can:
> 1. Use a voice-modem as FXO?

This is discussed at considerable length in the archives, wiki ad
nauseum.  
Short answer, no.

> 2. Use a Digium X100P?  What's the advantage over using a voice-modem?

Search archives about full duplex.  Voice modems don't.

> 3. Set up * as a SIP client for VoicePulse and sipgate.de.  I could 
> add lines via broadvoice.com and FWD?  And if I'm REALLY lucky, I 
> could even convince Vonage to allow open access and connect directly?

Yes and no.  Sipgate.de, voicepulse, iconnecthere, FWD and others yes.  
Vonage no.

> 4. Is anyone running an ATA186 into an FXO device?  Sound-quality?

It works fine, but I'm not keen on it either

>
> Are there multi-FXO cards, because I'm afraid I'll be running out of 
> PCI slots.

Digium is closer than ever to having its FXO modules to put on the
TDM400P.  
Stay tuned to www.digium.com

>
>
> To get my extensions connected, I can:
> 1. Use a Digium TDM400P?
> 2. Use one ore more Sipuras?
> 3. Use any Software IP Phone?
> 4. Use any Hardware IP Phone?

All of the above and several others including T100Ps and channel banks

>
> TDM400P cost $75/port, while the Sipuras are only $50/port.  Is there 
> an advantage to using the Digium?

The smart ass answer is Asterisk.

>
>
> Now once everything connected, it'll probably take me a while to get 
> things configured.  I assume that I can do pretty much anything I 
> want, just as long as I have access to the sources.  Can I:
>
> 1. Set up auto-attendants based on the incoming phone line?  Based on 
> id of the caller?

Yes, requires some practice

> 2. Set up least-cost call routing?

We are configuring this for an international callback customer, now.

> 3. Have integrated dialing plans, such as --
>      1 xxx yyy nnnn = call outside line
>      011 .... = call internationally
>      * xxx = call extension xxx
>      # 4 xxx yyy nnnn = call using outside line #4
>      911 = call 911 using actual landline
>    (My wife needs to be able to use this too)

At the risk of being even more than too cute, yes.

>
> From some of your who have set up and are maintaining * PBXs, how 
> difficult is it to get started for someone who doesn't do linux 8 
> hours a day (I'm a PC guy, but am maintaining a dedicated linux server

> for webhosting).

It is involved.  Mind you we, VCCH, are selling our services for just
sucn a 
situation.  We are delighted to teach as well as serve.

>
> What's the preferred linux distro for running Asterisk?  I have RH8 
> and RH9 here.

That will work fine and there are some documents hanging about to deal
with a 
variety of "RedHatisms".  We tend to use Mandrake (out of historical 
preference).  All in all, we are completely agnostic, well except for
FreeBSD 
(drivers).

>
> I think that's all -- thanks in advance for your help & answers!
>
> -- Jay

And about your Adtrans 750 question.  The list has reported a series of
issues 
with same.  I have an Adtrans 750 in my house with both FXO and FXS
cards.  
The Adtrans is connected to Asterisk via T1 (T100P) and works great.  We
did 
have Digium help us tweak it so caller ID worked more reliably.

I would not connect Asterisk to a channel bank anyway but T1.

Howard White
president - VCCH, Inc.		[Digium reseller and consultancy]
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