[asterisk-users] Sipura SPA3000

sdcharly at gmail.com sdcharly at gmail.com
Sat Sep 2 10:28:43 MST 2006


Hi all,

Im quite new to SPA3000.

I have a TRIXBOX running on public address. I need my SPA3000's FXO to be
used as a trunk from a dynamic address behind NAT. Is this scenario
possible?

Please give me some good links if it works.. I would really appreciate any
help as my TRIXBOX is in US and my SPA3000 in middle east.

Thanks everyone

Dan.



On 02/09/06, Rich Adamson <radamson at routers.com> wrote:
>
> That option addresses what to do with the fxs (line 1) when the
> registration fails as opposed to what does the fxo (pstn line) does when
> registration fails.
>
>
> Bob Chiodini wrote:
> > Rich,
> >
> > After reading a little more, how about the "Line 1 VoIP Fallback to
> > PSTN" (section 4.9)?  It looks like this is invoked when the Ethernet
> > link is down or registration fails.  I don't have a SPA3000 up at the
> > moment to look at what's required.
> >
> > Bob...
> >
> > On Fri, 2006-09-01 at 11:45 -0500, Rich Adamson wrote:
> >> If "pstn call ring thru line 1" is enabled, all incoming pstn calls
> will
> >> ring through to the fxs port (and not to asterisk). The OP was looking
> >> for a auto fail over function that essentially would be "pstn call ring
> >> thru line 1 on sip failure". That doesn't exist.
> >>
> >>
> >> 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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