[Asterisk-Users] spouse-friendly spa-3000 pstn interface

[Rich Adamson]

This post is simply documenting a spouse-friendly way of using the
spa-3000 as both a fxs and fxo port for basic soho environments in
the US, allowing asterisk to participate as needed/wanted.

All home phones are connected _only_ to the spa-3000 fxs port.

The incoming home pstn line is connected _only_ to the spa-3000 
fxo port.

Defined Line 1 (fxs) to register with asterisk via sip (extn 1111), 
with silence suppression disabled.

Defined the PSTN Line (fxo) to register with asterisk via sip using 
a second sip.conf entry (extn 2222).

PSTN User, defined PSTN Ring Thru Line 1 Ring Settings as "1".

In Line 1, defined Gateway Account #1 to point to asterisk, and
created a dialplan entry like:
 (*xx|81xxx.<:@gw1>|3xxx<:@gw1>|0<:@gw0>|[2-9]11<:@gw0>|xxxxxx.<:@gw0>)
Note: gw0 defaults to the pstn line per the spa-3000 doc.

Result:
1. If asterisk is down for any reason, all incoming pstn home calls 
still ring through to the analog house phones.
2. Incoming pstn calls ring through to the house phones without any
asterisk involvement, and callerid is properly passed to the house
phones. (Since my * also has an fxo port attached behind the spa-3000,
* still rings and I can answer the call via it.)
3. Loss of commercial AC power causes the spa-3000 to physically
cut through the pstn line to the house phones, leaving all house
phones operative. Same with loss of an ethernet physical connection.
4. Calls originating from within asterisk to the house phones use:
 exten => 1111,1,SetVar(ALERT_INFO=bellcore-r3)
 exten => 1111,2,Dial(SIP/1111,25,r)
 exten => 1111,3,Hangup
and ring the house phone with distinctive rings. Spouce recognizes
incoming pstn calls as different from voip/asterisk calls. (We have
about six analog phones behind the spa-3000 that properly ring on 
all calls.)
5. Calls placed from the house phones (via fxs Line 1) use the above
dialplan.
 a. LD calls starting with 81xxx. are routed via * to an ITSP (gw1)
 b. normal US local calls routed via pstn line (gw0)
 c. calls to 3xxx are asterisk extensions and are therefor routed
    to asterisk (gw1)
 d. 911 (as well as 211, 311, etc) are routed via the pstn line.
 e. 0 calls are routed via pstn line
If spouse is having a problem with dialing LD via 81xxx, then dialing
the same call without the 8 routes the LD call via pstn line.
6. Calls originating from within asterisk can be routed to this pstn
line via Dial(Sip/2222/...) if needed.
7. The old answering machine (that my spouse knows how to use) still
accepts all unanswered calls.

I fully understand the above approach keeps asterisk out of the loop
most of the time, and that was an objective (from a spouse perspective
for now). It represents the least intrusive integration without
remedial training, etc. :)  No need to purchase six sip phones either.

There has been no echo or any other negative issues thus far; haven't
tested a lot of things as yet though. (Eg, what happens when the house
phone is talking via voip conversation and an incoming pstn call
arrives? etc.) The spa-3000 is running v2.0.1(GWd).

Rich