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

Mon Sep 6 09:24:05 MST 2004

Rich,

I have an SPA-3000 with a similar setup to yours.  Unlike yours, my cordless
phone system is on a SPA-2000 that connect through the SPA-3000 for a PSTN
connection. For LD I use 91xxx for (MCI) or 71xxx for (VoicePulse).  In
addition to trunk access codes 7 & 9, I use FWD by dialing 8xxxxxx.

I also recently replaced our 2.4GHz phones with 5.8GHz phones.  I picked the
Uniden TRU8866 5.8GHz cordless system (I highly recommend it).  It supports
up to 10 handsets on one base unit and has excellent audio quality.  What
really sold me on this cordless phone system, besides the audio quality, was
the fact that it supports two lines.  I have connected both lines to the
SPA-2000 which allows us to utilize the local PSTN and VoIP trunk
simultaneously.

I like your post because I have also gone to great lengths to integrate the
* system into our home without alienating my wife or visiting friends &
family.  Fortunately I have been able to talk my wife into using the *
voicemail functions.  

At first my wife was not too keen on the VM system.  It was the Uniden sets
that sold her on it.  The Uniden system supports Message Waiting Indicator
(MWI) when messages are left on the * system.  So she can see from any
handset (red slow flashing LED) that a VM message is waiting.  The phone
base and handsets also have a feature button marked for checking VM.  So I
programmed the button on all the cordless handsets to dial the VM extension
2500#, which then prompts for the mailbox password.  So far so good, no echo
and the clarity is excellent whether talking to a PSTN or VoIP call.

The only problem I have not been able to figure out is flashing to the PSTN
line for call-waiting... It does not work and I get secondary dial tone
instead of the waiting call.  Any Ideas?

--Larry

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Rich Adamson
Sent: Monday, September 06, 2004 9:38 AM
To: Asterisk-a-users-list
Subject: [Asterisk-Users] spouse-friendly spa-3000 pstn interface

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".

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

_______________________________________________
Asterisk-Users mailing list
Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users