[asterisk-users] Sipura SPA3000

[Larry Alkoff]

Bob I have a further question about Fallback:

On my Line 1 tab, the last item is
   "VoIP Fallback To PSTN"
but there is no setting that can be changed.

I _think_ my firmware is the latest when I bought the unit Oct 25, 2005
but _possibly_ I have no fallback.  Bummer.  Could you comment on this 
please?

BTW, section 4.11 talks only about pstn calls ringing line 1:

"The voice path is (7) (6) (4) (2) (1). This feature is enabled by 
setting <PSTN Ring Thru Line 1>
to “yes”. If enabled, all incoming PSTN calls will ring the Line 1 phone 
regardless the VoIP gateway
function is enabled on the SPA or not. Hence the same phone can be used 
to receive calls from Line
1 VoIP and from the PSTN."


Section 4.9 talks about Fallback to PSTN but I'm not sure how to test 
this with my setup or even if fallback is implemented in my SPA3k:

"4.9. Line 1 VoIP Fallback to PSTN
When power is removed from the SPA-3000, the FXS port will be connected 
to the FXO port. In this
case, the telephone attached to the FXS port is electrically connected 
to the PSTN service via the
FXO port. When power is applied to the SPA, the FXS port will be 
disconnected from the FXO port.
However, if the PSTN line is in use when the power is applied to the 
SPA, the relay will not be flipped
until the PSTN line is released. This is done so that the SPA will not 
interrupt any call in progress on
the PSTN line.
When Line 1 VoIP service is down (due to registration failure or loss of 
Ethernet link), SPA can be
configured to automatically route all outbound calls to the internal 
gateway if <Auto PSTN Fallback>
([Line 1] tab) is set to “yes”. The PSTN gateway applies the <Line 1 
Fallback DP> to further limit the
calls that can be made by the Line 1 caller during the fallback 
operation; this dial plan may be set to
“none”. This case also belongs to call type #7 and the voice path is (1) 
(2) (4) (6) (7)."

Of course, I'm having a lot of trouble reading this complex manual <g>

Larry


Bob Chiodini wrote:
> Probably the PSTN Call Ring Thru Line 1 feature.  Section 4.11 in:
> 
> http://www.sipura.com/Documents/SipuraSPAUserGuidev2.0.9.pdf#search=%22spa3000%20manual%22
> 
> By default, if my asterisk went down after the SPA3000 was already
> registered, the in-bound PSTN call was lost.  I probably did not wait
> long enough and I did not have "PSTN Call Ring Thru Line 1" enabled.
> 
> Bob...
> 
> On Fri, 2006-09-01 at 10:32 -0400, Kevin Collins wrote:
>> My 3000 does this natively without config. 
>>
>>
>> Kevin Collins
>>  
>>
>> -----Original Message-----
>> From: asterisk-users-bounces at lists.digium.com
>> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve Kennedy
>> Sent: Friday, September 01, 2006 10:03 AM
>> To: asterisk-users at lists.digium.com
>> Subject: Re: [asterisk-users] Sipura SPA3000
>>
>> On Fri, Sep 01, 2006 at 08:11:50AM -0500, Rich Adamson wrote:
>>
>>>>   I have a question on configuration of SPA3000 with asterisk.
>>>>   1. I want all incoming calls are redirected from SPA3000 to my
>>>>      asterisk server.
>>>>   2. Asterisk then should direct this call to my SIP phones (including
>>>>      Sipura)
>>>>   3. In case asterisk server is down I want that call be directed
>>>>      straight to the handset connected to the Sipura Is this 
>>>> configuration possible?
>>> The spa3000 does not have logic in it to support #3.
>> I thought the SPA3K could do this, i.e. on power failure or non-ability to
>> connect to server, connect FXS to FXO.
>>
>>
>> Steve
>>
>> --
> 
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-- 
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux