[Asterisk-Users] spa 3000 help

[Rich Adamson]

> Arrggghh. Tearing my hair out here.
> 
> I'm trying to set up the spa3000 in the UK for my home, and want * to
> control the dial plan
> 
> I've googled to no avail. I've read the manual to no avail.
> 
> Can someone, please let me know what the parameters is the spa and * are to
> 
> a) receive a call from the pstn
> b) make a call to the pstn from the phone attached
> 
> I can make sip to sip calls (i.e. I can use xlite to call the phone, and the
> phone to call xlite)

For calls initiated from Line1, create a dialplan under the Line1 tab that looks
something like the following:
 (81xxx.<:@1.2.3.4;usr=3020;pwd=mysecret>|3xxx<:@gw1>|0<:@gw0>|[2-9]11<:@gw0>|xxxxxx.<:@gw0>)
which means:
 - if the user dials 8-1-xxx-xxxx, the call is routed to asterisk
 - if the user dials 3xxx, route the call to gw1 (which is defined also as asterisk)
 - if the user dials 0, route to the pstn port
 - if the user dials 911, 411, etc, route to the pstn
 - otherwise if no match, send the call to the pstn

One item (of several) that aren't very clear from the documentation is the use of
gw0, gw1, etc, within the spa3k dialplans. For the line1 dialplan, gw0 defaults to
the pstn port. Therefore in the above dialplan, 0<:@gw0 sends the call to the pstn
port. You'll notice two different ways in the above dialplan to send calls to *.

Within the spa3k, configure Line1 to register with asterisk and, under the pstn tab,
configure that interface to register with asterisk. These are two completely different
registrations and should have different User ID entries (eg, extn 1111 and 2222).
Once the two entries are properly entered, check using * cli with 'sip show registry'
to ensure the registrations are working as expected.

Calls from asterisk to Line1 use a standard Dial(Sip...) dialplan within asterisk.
Calls routed from asterisk to the PSTN port use the standard Dial(Sip...) entry as
well, using the appropriate UserID for (eg, 1111 or 2222) entered in the spa3k
registration above.

Due to the way I'm using the spa3k, I have all incoming pstn calls ring the line1
without passing through asterisk. So, not sure what parameters are used to direct
those calls to asterisk instead. My understanding is that others have done this.

Someone on this list noted that calls originating from asterisk and sent out via the
pstn interface are now failing. My implementation test of this about a week ago
suggests something changed within asterisk that precludes sending the dialed number
to the spa3k, but I've not had any time to trace the issue to identify the root cause.
It is entirely possible my asterisk config needs a little tweaking; not sure yet.

For my implementation, incoming pstn calls ring the phones attached to line1. If
asterisk sends a call to line1, I've configured the spa3k to provide distinctive
ringing to provide some indication where the incoming calls are coming from. Works
great.

FWIW, I'm running cvs head from mid September along with the latest spa3k firmware.
I do have some echo issues (as others have) with the spa3k and apparently Sipura
is working on that (based on their release notes).

Rich