RE: [Asterisk-Users] Has anyone tried using a Sipura-3000 as an FXO device for *?

[Dameon D. Welch-Abernathy]

On Sat, 2004-07-31 at 17:43, Kevin Walsh wrote:

> As I said, I don't have one of these yet.  Do you happen to know what
> the box would do if the dialplan said to route the call to <:@gw0>
> and that port was already in use?

You'd probably get a Fast Busy if dialing from the FXS port. If coming
in from Asterisk, I think you'd get the appropriate SIP message saying
the line was busy (503?)

-- PhoneBoy

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RE: [Asterisk-Users] Has anyone tried using a Sipura-3000 as an FXO device for *?

[Rich Adamson]

> Rich Adamson [EMAIL PROTECTED] wrote:
> > Kevin Walsh wrote:
> > > You can apparently use the SPA-3000 dialplan to specify that the
> > > call should go via its FXO port, without going via Asterisk.  This
> > > could be useful for emergency services.  I don't have a SPA-3000 yet,
> > > so I can't say what happens if you try to route an emergency call via
> > > the FXO port and that port is in use.  Perhaps it sends the call to
> > > Asterisk instead.  I'll find out when I get mine and play with it.
> > >
> > Yes, the dialplan for the fxs line can look like:
> >  (*xx|[34569]11<:@gw0>|0|00|[2-9]xx<:@gw0>|1xxx[2-9]xxS0|.)
> > where 911 is sent to gw0 (the fxo port),
> > calls to Nxx (local calls) go to gw0,
> > and 1+ calls (long distance) go to a voip box (* in my case)
> > 
> As I said, I don't have one of these yet.  Do you happen to know what
> the box would do if the dialplan said to route the call to <:@gw0>
> and that port was already in use?

The purpose for mentioning the above was simply to indicate the dialplan
includes a fair amount of flexibility.

> If the call simply fails then that's a wasted facility, and I wouldn't
> use it;  If Asterisk was in charge then it could loop the call back to
> the FXO or sent it via another route.  I suspect that the SPA would
> try the <:gw0> first and then fall back to the SIP link, either
> automatically or as an option, before giving up.

The options are there to allow that config as well. It's totally up
to the individual configuring the box and not artifically limited
by the manufacturer. However, I've not seen anything as yet that
would allow 'if "a" is down/busy, use a secondary "b" for call
completion'.

> > Rather sophisticated little box. :)
> >
> So it would seem.  I can't wait to get my hands on one in September.



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RE: [Asterisk-Users] Has anyone tried using a Sipura-3000 as an FXO device for *?

[Kevin Walsh]

Rich Adamson [EMAIL PROTECTED] wrote:
> Kevin Walsh wrote:
> > You can apparently use the SPA-3000 dialplan to specify that the
> > call should go via its FXO port, without going via Asterisk.  This
> > could be useful for emergency services.  I don't have a SPA-3000 yet,
> > so I can't say what happens if you try to route an emergency call via
> > the FXO port and that port is in use.  Perhaps it sends the call to
> > Asterisk instead.  I'll find out when I get mine and play with it.
> >
> Yes, the dialplan for the fxs line can look like:
>  (*xx|[34569]11<:@gw0>|0|00|[2-9]xx<:@gw0>|1xxx[2-9]xxS0|.)
> where 911 is sent to gw0 (the fxo port),
> calls to Nxx (local calls) go to gw0,
> and 1+ calls (long distance) go to a voip box (* in my case)
> 
As I said, I don't have one of these yet.  Do you happen to know what
the box would do if the dialplan said to route the call to <:@gw0>
and that port was already in use?

If the call simply fails then that's a wasted facility, and I wouldn't
use it;  If Asterisk was in charge then it could loop the call back to
the FXO or sent it via another route.  I suspect that the SPA would
try the <:gw0> first and then fall back to the SIP link, either
automatically or as an option, before giving up.

>
> Rather sophisticated little box. :)
>
So it would seem.  I can't wait to get my hands on one in September.

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RE: [Asterisk-Users] Has anyone tried using a Sipura-3000 as an FXO device for *?

[Rich Adamson]

> Wolfgang S. Rupprecht [EMAIL PROTECTED] wrote:
> > [EMAIL PROTECTED] (Rich Adamson) writes:
> > > Like *, it also has an internal dialplan, however understanding the
> > > various interactions requires some experimentation, as each of the
> > > interfaces seem to be considered a "gateway", and part of the dialplan
> > > directs calls to gw0, gw1, gw2 (etc) which correspond to physical
> > > interfaces in most cases.
> > >
> > I felt some pangs of guilt turning all that stuff off, but I couldn't
> > think of any time I'd want two dialplans in series.
> > 
> It saves having to wait for an inter-digit timeout to expire when
> dialling via the FXS port.  The SPA-[123]000 dialplan will recognise
> your dial string and send it immediately to Asterisk (if so configured).
> Asterisk will take it from there.
> 
> You can apparently use the SPA-3000 dialplan to specify that the
> call should go via its FXO port, without going via Asterisk.  This
> could be useful for emergency services.  I don't have a SPA-3000 yet,
> so I can't say what happens if you try to route an emergency call via
> the FXO port and that port is in use.  Perhaps it sends the call to
> Asterisk instead.  I'll find out when I get mine and play with it.

Yes, the dialplan for the fxs line can look like:
 (*xx|[34569]11<:@gw0>|0|00|[2-9]xx<:@gw0>|1xxx[2-9]xxS0|.)
where 911 is sent to gw0 (the fxo port),
calls to Nxx (local calls) go to gw0,
and 1+ calls (long distance) go to a voip box (* in my case)

The above is from a test spa3000 that is not in production, so the
actual dialplan is not complete as yet. The fxs dialplan is limited 
to a single line, and can only have a limited number of characters 
in that line.

There are an additional 8 dialplans that appear to be oriented
around how to deal with incoming pstn calls, and routing those
to * (or whatever). Haven't played with these at all as yet.
The doc suggests these 8 dialplans can also be tied in with user
assigned pin numbers, allowing a user to call into the spa3000
via the pstn, enter their pin, and be routed to * (or different
voip providers).

Rather sophisticated little box. :)

Rich



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RE: [Asterisk-Users] Has anyone tried using a Sipura-3000 as an FXO device for *?

[Kevin Walsh]

Wolfgang S. Rupprecht [EMAIL PROTECTED] wrote:
> [EMAIL PROTECTED] (Rich Adamson) writes:
> > Like *, it also has an internal dialplan, however understanding the
> > various interactions requires some experimentation, as each of the
> > interfaces seem to be considered a "gateway", and part of the dialplan
> > directs calls to gw0, gw1, gw2 (etc) which correspond to physical
> > interfaces in most cases.
> >
> I felt some pangs of guilt turning all that stuff off, but I couldn't
> think of any time I'd want two dialplans in series.
> 
It saves having to wait for an inter-digit timeout to expire when
dialling via the FXS port.  The SPA-[123]000 dialplan will recognise
your dial string and send it immediately to Asterisk (if so configured).
Asterisk will take it from there.

You can apparently use the SPA-3000 dialplan to specify that the
call should go via its FXO port, without going via Asterisk.  This
could be useful for emergency services.  I don't have a SPA-3000 yet,
so I can't say what happens if you try to route an emergency call via
the FXO port and that port is in use.  Perhaps it sends the call to
Asterisk instead.  I'll find out when I get mine and play with it.

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Re: [Asterisk-Users] Has anyone tried using a Sipura-3000 as an FXO device for *?

[Rich Adamson]

> > The box was truly targeted for the residential user where existing
> > phones interface on one side, the pstn line on the other side, and
> > the default call is sent to the voip interface. Disconnected (or
> > failed) ethernet results in a relay flipping, tying the fxs directly
> > to the fxo. Same with power failure. Nice.
> 
> I think the cut-through from the fxs to the fxo (and backwards) is via
> a digital connection.  In normal use you appear to end up getting hit
> by the digitization delays.  As far as I can tell the relay
> cut-through is only used for power failure.

It's actually a relay, and you can hear/feel it. The cut-through actually 
works by either removing power, or, removing the cat5 cable. However, 
it wouldn't have a clue whether a layer-3 box (including *) were down.

> > Initial tests did not show any signs of echo, very good volume and 
> > audio quality, and would probably be a good choice for small quantities
> > of pstn lines (particularily soho and residential users).
> 
> I still notice some low-volume problems with
> FXO->asterisk->grandstream-bt101 even though I bumped the FXO incoming (and
> outgoing) gains to +12dB.  (To keep calls from the FXO->asterisk->FXS
> a reasonable volume I needed to drop the gain of the fxs port to -15
> (from the factory of -3).
> 
> Somebody with a real phone VU meter needs to have a look at the
> Sipura-3000 FXO.  I can't believe it is off that much.  Might the
> Grandstream BT-101 be really low in volume and I'm just mistakenly
> blaming the volume problem on the Sipura?

That's odd; sort of sounds like BT101 problem. Using C7960's, the 
volume was excellent (without touching anything). Using an analog
set on the fxs port was very "hot", and dropping the fxs gain slightly
improved that to what a non-technical user would suggest is normal.
(I did use a $3500 transmission test set on as well.)
 
> Given the choice between hearing dead air and hearing the tones, I
> think I'd rather hear the tones.  At least I know something is
> happening.

I'd suspect that non-technical users would raise a small issue with the
tone feedback (at least in the US), as their not acustomed to hearing
that on normal calls.

Rich


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Re: [Asterisk-Users] Has anyone tried using a Sipura-3000 as an FXO device for *?

[Wolfgang S. Rupprecht]

[EMAIL PROTECTED] (Rich Adamson) writes:
> Like *, it also has an internal dialplan, however understanding the
> various interactions requires some experimentation, as each of the
> interfaces seem to be considered a "gateway", and part of the dialplan
> directs calls to gw0, gw1, gw2 (etc) which correspond to physical
> interfaces in most cases.

I felt some pangs of guilt turning all that stuff off, but I couldn't
think of any time I'd want two dialplans in series.

> The box was truly targeted for the residential user where existing
> phones interface on one side, the pstn line on the other side, and
> the default call is sent to the voip interface. Disconnected (or
> failed) ethernet results in a relay flipping, tying the fxs directly
> to the fxo. Same with power failure. Nice.

I think the cut-through from the fxs to the fxo (and backwards) is via
a digital connection.  In normal use you appear to end up getting hit
by the digitization delays.  As far as I can tell the relay
cut-through is only used for power failure.

> Initial tests did not show any signs of echo, very good volume and 
> audio quality, and would probably be a good choice for small quantities
> of pstn lines (particularily soho and residential users).

I still notice some low-volume problems with
FXO->asterisk->grandstream-bt101 even though I bumped the FXO incoming (and
outgoing) gains to +12dB.  (To keep calls from the FXO->asterisk->FXS
a reasonable volume I needed to drop the gain of the fxs port to -15
(from the factory of -3).

Somebody with a real phone VU meter needs to have a look at the
Sipura-3000 FXO.  I can't believe it is off that much.  Might the
Grandstream BT-101 be really low in volume and I'm just mistakenly
blaming the volume problem on the Sipura?

> The only downside I've seen thus far (not much experience as yet) is
> that * calls to the pstn line are cut through immediately, so one 
> hears the initial dialtone from the pstn and the sending of the dtmf
> tones on all outgoing calls. Kind of annoying, but there might be 
> some config option to handle it; I've just not found it as yet. (If
> anyone knows how to handle that, sure would appreciate a suggestion.)

Given the choice between hearing dead air and hearing the tones, I
think I'd rather hear the tones.  At least I know something is
happening.

-wolfgang
-- 
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch
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RE: [Asterisk-Users] Has anyone tried using a Sipura-3000 as an FXO device for *?

[DUSTIN WILDES]

I found it was worse when using the G726 or G723 codecs, but if you used the G711 
codec, the DTMF echo was hardly noticable.  I was using the latest image:  2.0.9d



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rich Adamson
Sent: Wednesday, July 28, 2004 8:31 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Has anyone tried using a Sipura-3000 as an
FXO device for *?


> On Tue, 2004-07-27 at 15:52, Carmi Weinzweig wrote:
> > I am considering using Sipura-3000s as FXO devices for my * system. Has 
> > anyone tried them in that configuration? They interest me because they 
> > need no PCI slots and therefore no drivers. I would much prefer not to 
> > have any special kernel requirements for my system.
> 
> A number of us are using SPA-3000s for this exact purpose, including
> myself. Works pretty well.
> 
> -- PhoneBoy

Have you found a way to get rid of the dial tone and dtmf tones when
placing an outbound pstn call through the 3000?

In my config, the call completes as expected however the dialtone and
dtmf tones are slightly annoying.

Rich


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Re: [Asterisk-Users] Has anyone tried using a Sipura-3000 as an FXO device for *?

[Rich Adamson]

> On Tue, 2004-07-27 at 15:52, Carmi Weinzweig wrote:
> > I am considering using Sipura-3000s as FXO devices for my * system. Has 
> > anyone tried them in that configuration? They interest me because they 
> > need no PCI slots and therefore no drivers. I would much prefer not to 
> > have any special kernel requirements for my system.
> 
> A number of us are using SPA-3000s for this exact purpose, including
> myself. Works pretty well.
> 
> -- PhoneBoy

Have you found a way to get rid of the dial tone and dtmf tones when
placing an outbound pstn call through the 3000?

In my config, the call completes as expected however the dialtone and
dtmf tones are slightly annoying.

Rich


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Re: [Asterisk-Users] Has anyone tried using a Sipura-3000 as an FXO device for *?

[Rich Adamson]

Only for basic testing. By default, incoming pstn calls ring the fxs
line. However, there is an option to disable that and apparently route
the call to the voip system. There is apparently another option that
involves a timeout, routing the call to * if the fxs doesn't answer
within the timeout period. I've not played with those options as yet.


> I am most interested in using it for incoming calls. Have you tried 
> that yet?
> 
> /carmi
 
> On Jul 27, 2004, at 5:30 PM, Rich Adamson wrote:
> 
> >
> >> I am considering using Sipura-3000s as FXO devices for my * system. 
> >> Has
> >> anyone tried them in that configuration? They interest me because they
> >> need no PCI slots and therefore no drivers. I would much prefer not to
> >> have any special kernel requirements for my system.
> >
> > In the process of doing that now.
> >
> > Simple / prelim implementation:
> >
> > Each of the three ports (eg, fxs, fxo, cat5) are treated as separate
> > interfaces, and one can configure fxo -> *, fxs -> *, ring-through from
> > fxo -> fxs, * g/w functions to the pstn, etc. There seems to be a ton
> > of functionality in the box and those functions are mostly limited by
> > your imagination (and how well one can read and comprehend).
> >
> > Configurable from a web interface, however there are a ton of options
> > that aren't very clear without digging deep into their newly released
> > admin manual (called a user guide on their site). The manual seems to
> > have been written for the 1000/2000 with additional chapters/sections
> > oriented to the 3000. (Sort of rush to print.)
> >
> > The fxo and fxs interfaces can be configured to register separately
> > with *, making both very addressable, etc.
> >
> > Like *, it also has an internal dialplan, however understanding the
> > various interactions requires some experimentation, as each of the
> > interfaces seem to be considered a "gateway", and part of the dialplan
> > directs calls to gw0, gw1, gw2 (etc) which correspond to physical
> > interfaces in most cases.
> >
> > The box was truly targeted for the residential user where existing
> > phones interface on one side, the pstn line on the other side, and
> > the default call is sent to the voip interface. Disconnected (or
> > failed) ethernet results in a relay flipping, tying the fxs directly
> > to the fxo. Same with power failure. Nice.
> >
> > So, properly configured, it appears to be a very nice box that would
> > allow * to sit in the middle, but still provide excellent fail-over
> > capabilities when unusual events occur.
> >
> > For small installations, it makes handling US 911 calls extremely
> > easy as that can be made part of the internal dialplan.
> >
> > Initial tests did not show any signs of echo, very good volume and
> > audio quality, and would probably be a good choice for small quantities
> > of pstn lines (particularily soho and residential users).
> >
> > The only downside I've seen thus far (not much experience as yet) is
> > that * calls to the pstn line are cut through immediately, so one
> > hears the initial dialtone from the pstn and the sending of the dtmf
> > tones on all outgoing calls. Kind of annoying, but there might be
> > some config option to handle it; I've just not found it as yet. (If
> > anyone knows how to handle that, sure would appreciate a suggestion.)
> >
> > Thus far, I'd give the box at least an A-, and will likely move
> > higher with a little more experience.
> >
> > Rich
> >
> >
> >
> > ___
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---End of Original Message-


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Re: [Asterisk-Users] Has anyone tried using a Sipura-3000 as an FXO device for *?

[Dameon D. Welch-Abernathy]

On Tue, 2004-07-27 at 15:52, Carmi Weinzweig wrote:
> I am considering using Sipura-3000s as FXO devices for my * system. Has 
> anyone tried them in that configuration? They interest me because they 
> need no PCI slots and therefore no drivers. I would much prefer not to 
> have any special kernel requirements for my system.

A number of us are using SPA-3000s for this exact purpose, including
myself. Works pretty well.

-- PhoneBoy

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Re: [Asterisk-Users] Has anyone tried using a Sipura-3000 as an FXO device for *?

[Greg Broiles]

I have a Sipura-3000 and am hoping to use it to provide FXS/FXO ports
for my Asterisk box. I don't have it working well yet but I blame that
on my inexperience with Asterisk. Some configuration examples are
available at 

or .

-- 
Greg Broiles, JD, EA
[EMAIL PROTECTED] (Lists only. Not for confidential communications.)
Law Office of Gregory A. Broiles
San Jose, CA
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Re: [Asterisk-Users] Has anyone tried using a Sipura-3000 as an FXO device for *?

[Carmi Weinzweig]

I am most interested in using it for incoming calls. Have you tried 
that yet?

/carmi
On Jul 27, 2004, at 5:30 PM, Rich Adamson wrote:

I am considering using Sipura-3000s as FXO devices for my * system. 
Has
anyone tried them in that configuration? They interest me because they
need no PCI slots and therefore no drivers. I would much prefer not to
have any special kernel requirements for my system.
In the process of doing that now.
Simple / prelim implementation:
Each of the three ports (eg, fxs, fxo, cat5) are treated as separate
interfaces, and one can configure fxo -> *, fxs -> *, ring-through from
fxo -> fxs, * g/w functions to the pstn, etc. There seems to be a ton
of functionality in the box and those functions are mostly limited by
your imagination (and how well one can read and comprehend).
Configurable from a web interface, however there are a ton of options
that aren't very clear without digging deep into their newly released
admin manual (called a user guide on their site). The manual seems to
have been written for the 1000/2000 with additional chapters/sections
oriented to the 3000. (Sort of rush to print.)
The fxo and fxs interfaces can be configured to register separately
with *, making both very addressable, etc.
Like *, it also has an internal dialplan, however understanding the
various interactions requires some experimentation, as each of the
interfaces seem to be considered a "gateway", and part of the dialplan
directs calls to gw0, gw1, gw2 (etc) which correspond to physical
interfaces in most cases.
The box was truly targeted for the residential user where existing
phones interface on one side, the pstn line on the other side, and
the default call is sent to the voip interface. Disconnected (or
failed) ethernet results in a relay flipping, tying the fxs directly
to the fxo. Same with power failure. Nice.
So, properly configured, it appears to be a very nice box that would
allow * to sit in the middle, but still provide excellent fail-over
capabilities when unusual events occur.
For small installations, it makes handling US 911 calls extremely
easy as that can be made part of the internal dialplan.
Initial tests did not show any signs of echo, very good volume and
audio quality, and would probably be a good choice for small quantities
of pstn lines (particularily soho and residential users).
The only downside I've seen thus far (not much experience as yet) is
that * calls to the pstn line are cut through immediately, so one
hears the initial dialtone from the pstn and the sending of the dtmf
tones on all outgoing calls. Kind of annoying, but there might be
some config option to handle it; I've just not found it as yet. (If
anyone knows how to handle that, sure would appreciate a suggestion.)
Thus far, I'd give the box at least an A-, and will likely move
higher with a little more experience.
Rich

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Re: [Asterisk-Users] Has anyone tried using a Sipura-3000 as an FXO device for *?

[Mike Benoit]

Yes, they seem to have many benefits. No PCI slots are used, echo issues
related to mainboards are eliminated, no interrupt sharing/dropping
issues, the list goes on.

My SPA-3000 was ordered last week and should be here any day now. Once I
do some testing on it I'll be sure to write a "review" comparing it
directly to Digium FXO cards for the mailing list. 

On Tue, 2004-07-27 at 18:52 -0400, Carmi Weinzweig wrote:
> I am considering using Sipura-3000s as FXO devices for my * system. Has 
> anyone tried them in that configuration? They interest me because they 
> need no PCI slots and therefore no drivers. I would much prefer not to 
> have any special kernel requirements for my system.
> 
> /carmi
> 
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Re: [Asterisk-Users] Has anyone tried using a Sipura-3000 as an FXO device for *?

[Rich Adamson]

> I am considering using Sipura-3000s as FXO devices for my * system. Has 
> anyone tried them in that configuration? They interest me because they 
> need no PCI slots and therefore no drivers. I would much prefer not to 
> have any special kernel requirements for my system.

In the process of doing that now.

Simple / prelim implementation:

Each of the three ports (eg, fxs, fxo, cat5) are treated as separate 
interfaces, and one can configure fxo -> *, fxs -> *, ring-through from
fxo -> fxs, * g/w functions to the pstn, etc. There seems to be a ton
of functionality in the box and those functions are mostly limited by
your imagination (and how well one can read and comprehend).

Configurable from a web interface, however there are a ton of options
that aren't very clear without digging deep into their newly released
admin manual (called a user guide on their site). The manual seems to
have been written for the 1000/2000 with additional chapters/sections
oriented to the 3000. (Sort of rush to print.)

The fxo and fxs interfaces can be configured to register separately
with *, making both very addressable, etc.

Like *, it also has an internal dialplan, however understanding the
various interactions requires some experimentation, as each of the
interfaces seem to be considered a "gateway", and part of the dialplan
directs calls to gw0, gw1, gw2 (etc) which correspond to physical
interfaces in most cases.

The box was truly targeted for the residential user where existing
phones interface on one side, the pstn line on the other side, and
the default call is sent to the voip interface. Disconnected (or
failed) ethernet results in a relay flipping, tying the fxs directly
to the fxo. Same with power failure. Nice.

So, properly configured, it appears to be a very nice box that would
allow * to sit in the middle, but still provide excellent fail-over
capabilities when unusual events occur.

For small installations, it makes handling US 911 calls extremely
easy as that can be made part of the internal dialplan.

Initial tests did not show any signs of echo, very good volume and 
audio quality, and would probably be a good choice for small quantities
of pstn lines (particularily soho and residential users).

The only downside I've seen thus far (not much experience as yet) is
that * calls to the pstn line are cut through immediately, so one 
hears the initial dialtone from the pstn and the sending of the dtmf
tones on all outgoing calls. Kind of annoying, but there might be 
some config option to handle it; I've just not found it as yet. (If
anyone knows how to handle that, sure would appreciate a suggestion.)

Thus far, I'd give the box at least an A-, and will likely move 
higher with a little more experience.

Rich



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