[Asterisk-Users] Questions on various and sundry IP phones, and cabling

[Adolf Osborne]

I'm attempting to do a first-time Asterisk install at home, firstly for 
use by my self and my family, and secondly as a learning experience.  
I've got a new house, and the previous owners removed all but one (1) 
phone jack.  So I figured I might as well build a PBX.

Functional goals include station-to-station calling, rudimentary auto 
attendant/voice mail, and perhaps tieing into the Altigen box that I've 
got at work via h.323.  But my first goal is to get any of my devices to 
talk to Asterisk, which so far I've been unable to do.

Hardware in-hand consists of a K6-2 based 2.6 kernel Gentoo box, two 
Selsius 30 VIP phones, a couple of Selsius 12SP+'s, a loaner Altigen 
IP-600 (which is nothing like a Polycom IP600, despite the name), and a 
switched 100 megabit network.  There's an X100P on order which should be 
here real soon now, which will be sharing the house's solitary CO line 
with a Uniden cordless phone until I get an FXS interface of some kind 
(which won't be until after the rest of the kit all works).

I've been reading list archives and the wiki pages for weeks, but now 
that I'm starting actual implementation, there's a few things that I'm 
not sure how to go about.

So.  The questions:

1.  Is the K6-2 meaty enough to do function in this enviroment?  I don't 
ever anticipate doing any hefty compression, as this is all in-house 
(unless I get the remote Altigen box flying, which will add one 
potential g.723 connection...).  Routing calls around between skinny 
phones and a single PSTN connection shouldn't be very taxing, should it?

2.  Asterisk seems to support the Selsius phones, maybe, via chan_sccp 
or chan_skinny.  I say "maybe" because, though some people on this list 
seem to have them working fine, documentation on any of it is extremely 
lacking.  The phones are -old-, booting to a 1997 copyright screen and 
reporting K2.01 as a version.  The results so far seem somewhat atypical 
of what's reported on this list.

Currently, I've got a 30 VIP configured with DHCP enabled, and have the 
TFTP server configured with Aterisk's address.  The phone boots, 
attempts to request SEPDefault.cnf and SEP0010EB001BCA.cnf (neither of 
which exist) via TFTP, then reboots a few minutes later and tries 
again.  Asterisk console with a dozen -v's and a -d doesn't show any 
activity during this time, whether using chan_skinny or chan_sccp.  
Behavior is the same with recent Asterisk CVS or the 0.9.0 ebuild in 
portage.  Standard behavior, as far as I can tell, is to at least see a 
flurry of registration attempts. 

Is there any way to better see what all is going on?

It's important that these phones work, because without them, there's no 
point in going any further with the project.  They were free.  :)  ISTR 
some information on Selsius phones specifically, lurking at Lambda 
Solutions, but it's all gone now.

3.  The IP-600.  For those unfamiliar, it's more-or-less a generic h.323 
endpoint, with the usual settings for gatekeeper and such, along with 
some Altigen extensions that are safe to ignore.  It does not speak SIP, 
or any other protocol.  In its native Altigen enviroment, it will only 
speak either g.711 and g.723.1 6.3k, but the phone itself might support 
additional codecs.  I've successfully used this phone by itself to call 
Netmeeting without a hitch.  Currently, my Asterisk build lacks h.323 
support, because neither the builtin h323 channel nor oh323 are willing 
to compile with CVS (and gentoo's non-cvs zaptel driver is unhappy with 
kernel 2.6), but I'll probably be able to make that end of things work 
on my own.

When I finally do get the an h.323 channel configured, what sort of 
behavior should I expect out of a default Asterisk install?  Should I be 
able to ring the server's IP and get back a generic auto attendant, or 
what?  In other words, how much configuring of Asterisk is necessary to 
see that h323 (or skinny, for that matter) is working between the phone 
and Asterisk and get audio going in both directions?  And, again, is 
there a better way to see what's going on than the messages in the 
Asterisk console?

4.  As I mentioned earlier, for the time being, Asterisk will be sharing 
it's phone line with a cordless phone.  Is the combination of the X100P 
card and Asterisk enough to detect when the cordless is off-hook, and 
speak an error message instead of stupidly attempting to dial out?  I 
intend to have Asterisk answer all incoming calls just after CID 
reception, and present an auto-attendant/telemarketer filter, with an 
option to immediately drop the call.  The desired behavior is thus as 
follows:

Inbound:  Asterisk answers after two rings and presents an auto 
attendant.  I elect to answer the call with the (non-*) cordless, hit 
(say) 8, and Asterisk hangs up and gets out of the way.  Else, keep the 
attendant going and act like a PBX.

Outbound:  When connecting to the PSTN, Asterisk first checks to see if 
the line is in use (by sensing loop voltage, or however it's normally 
done).  If in use, it presents a recording, or a reorder tone (or 
anything, really), and never goes off-hook.  If not in use, the call 
proceeds normally.

Are there any problems with this configuration?

Any advice, recommendations, or foolish commentary would be welcome.  
I'll be documenting and publishing the process of making the Selsius 
phones fly by the time it's all done, because there seems to be a lot of 
these phones out there still...

Thanks!