[asterisk-users] Sipura SPA3000

[Bob Chiodini]

Larry,

There is later firmware 3.1.10 dated March 2006.

I gave up on the SPA3K.  I could not solve the echo problems.

Rich Adamson indicated that the SPA3K did not have logic to fall back to
the PSTN on SIP failure, only loss of link to the network.  I would have
thought that Sipura could have used a registration failure as a SIP
failure.  The registration time would need to be relatively short to
detect a failure in a reasonable amount of time.

To test registration failover to PSTN, change the password for the SPA3K
on the asterisk side.  That should cause a registration failure.  Then
try placing a call to the PSTN line from elsewhere (a cell).  Try
calling out as well.

Neither Linksys nor Sipura seem to have updated manuals.  Try the latest
firmware, but get a copy of the current, just in case.

Bob...

On Wed, 2006-11-15 at 23:22 -0600, Larry Alkoff wrote:
> 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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