[sld]

For use in building a 4x4 line SOHO PBX, four Sipura SPA-3000s (Voxilla price $500) are less expensive than a Digium TDM40B 4-port FXS PCI card plus a TDM04B 4-port FXO card (Digium price $642). A similar case could be made for other FXO/FXS configurations using a combination of SPA-2000s and SPA-3000s.

Have any of you made this tradeoff of external Ethernet-connected ATAs vs. PCI FXS/FXO cards? Other than price, I would appreciate if anyone would share their view of the pros and cons of this approach.

Thanks!
-sld

[muppetmaster]

This is what I use for a SoHo system. Works fine. A couple of complaints though:

- During a one stage dial you can hear the 3000 pick-up and dial over the line, would be great to hear Asterisk ringing insted
- Echo/silence issues (see other posts on this forum) the line, hopefully these begin to be addressed in the pending 2.10 release

Other than that, works fine.

[PhoneBoy]

The "pick and dial over the line" issue is definately addressed in 2.0.10. I'm not 100% certain on the echo issues, though I've heard a slight improvement.

[gbroiles]

If you want to use some Asterisk features like meetme or IAX2 trunking, you need to have a hardware timer, which is included in the Digium cards. It might also be present on your USB controller. See http://www.voip-info.org/tiki-index....terisk%20timer.

Also, the multiple Sipura approach gives you that many AC adapters, Ethernet connections, etc, as well as more devices to configure/update/etc.

On the other hand, I have been using a 3000 to provide an FXS and FXO port to one of my Asterisk boxes and it's been working fine. I haven't had an issue with the dialout stuff because I don't dial out over the FXO port, and have been using VoIP outgoing instead.

[sk3-483]

If Sipura can get through to resolving the echo issue them I'll deploy more SPA3000. However, they have yet to acknowledge the problem. In my case they advised that I lower the gain settings and fiddle with the line impedance setup. They've been less than forthcomming with advice and assistance. It's dissappointing since I took part in the beat program and actually paid more for the device than it's final retail price.

I'm seriously considering the TDM400 as my next move. However, I've not had a good experience with X101p either.

Michael

[PhoneBoy]

Have you tried the latest firmware, which is supposed to help somewhat?

[sk3-483]

Have just installed the new firmware today. Will try it for a week and see what happens. I will note that oin order to get any reasonable level on my IP desk phone I had to set the PSTN>VOIP gain at +12. The echo was not as bad a previously, but I'll wait and see how it behaves when real calls come in.

Test calls from one side of the office to another are not really a good reference.

Michael

[TriRyche]

Any new updates/comments on this topic?

I am planning my Asterisk-based system at this time.

Here's what I have -

• House is now wired for 4 lines + ethernet per room
  One SPA-2000 (to be unlocked) from a previous provider
  DSL + 1 POTS line
  GIZMO 2.0 (with Asterisk support)
  3 SIP (or IAX) configured VOIP phone numbers from Sixtel

My choices now are to choose -

• additional SIP phones (i.e Linksys SPA942)
  a Wi-Fi handset (i.e. Linksys WIP 300/ WIP 330)
  additional Sipura adapters (SPA- ?)
  Digium TDM400P + TDM-FXS/FXO modules

I figure I should have 1 FXO adapter to connect to my POTS line.

I plan to use my SPA-2000 to -

• distribute VOIP 'line 1' throughout the house (for 2-3 analog phones)
  use VOIP 'line 2' to connect a fax machine

Hopefully, the SPA-2000 'line 1' can be configured for 2 incoming numbers (one with distinctive ring).
If not, I may need an additional FXS.

As for a Wi-Fi handset, I may wait until later in the year,
as Linksys is supposed to release some additional models in 2006.

I do not plan on purchasing any additional analog telephones.
Perhaps I do not need any Digium adapters if I plan to only purchase
SIP/Wi-Fi-based phones from here on in?

Maybe my question should be - Sipura or Digium for FXO duties?
(along with any other pros & cons for Digium(inboard) vs. Sipura(outboard) in general.


Here is another voxilla thread addressing similar concerns


- TriRyche

[mberlant]

Inboard vs. outboard is a personal preference based upon your priorities. Inboard will give you better fax handling via your PSTN line, while an SPA3000 (or -3102) will give you power failure coverage and the ability to locate the ATA physically apart from the Asterisk. I personally use the Sipura for these reasons, with the ATA in the wiring closet and the Asterisk server in my office upstairs.

As for your comment about putting two phone numbers on one SPA-2000 channel, could you elaborate on this? There are several ideas that come to mind, but they may not be what you envision. With each FXS device/channel having its own Asterisk extension number, you are completely free to route any of your inbound VoIP or PSTN services to any/all extensions in your system. While performing this routing, you can have extensions.conf set "alert-info" to invoke any of the 8 distinctive ring patterns available in the SPA.

Moreover, you can manipulate the incoming Caller ID to reflect the inbound channel of the phone call. For example, for any call coming in via a US VoIP service I prepend "1" to the CALLERID(number) and prepend a short code (like "IPK1234-" for IPKall line 1234) to the CALLERID(name). I can then use the phone's Call History function to return a call because it already has the 1 added in front and can also identify the incoming line because it is annotated before the caller's name on the CID display.

[TriRyche]

Originally Posted by mberlant

As for your comment about putting two phone numbers on one SPA-2000 channel, could you elaborate on this?

There are several ideas that come to mind, but they may not be what you envision. With each FXS device/channel having its own Asterisk extension number,
you are completely free to route any of your inbound VoIP or PSTN services to any/all extensions in your system. While performing this routing, you can have
extensions.conf set "alert-info" to invoke any of the 8 distinctive ring patterns available in the SPA.

Moreover, you can manipulate the incoming Caller ID to reflect the inbound channel of the phone call.
For example, for any call coming in via a US VoIP service I prepend "1" to the CALLERID(number) and prepend a short code (like "IPK1234-" for IPKall line 1234)
to the CALLERID(name). I can then use the phone's Call History function to return a call because it already has the 1 added in front and can also identify the incoming line
because it is annotated before the caller's name on the CID display.

Based on your reply, it sounds like I am able to accomplish what I want and more with the SPA-2000!

I know Asterisk can accomplish whatever scenario I can dream-up at the moment.
I am uncertain how many different incoming/outgoing providers/channels could be configured for 'line 1' and 'line 2' within the SPA-2000
and how elaborate a 'feature set' each channel can have available to itself. (apparently, it's quite the feature set, so let's focus on the
number of 'virtual lines' that can be handled by the SPA-2000 for now)

As a one-man operation for the most part, I do not have a current need to have more than 2 calls actually 'in progress' at any one time
(i.e. have the SPA-2000 'handling' a few calls on 'line 1,' (speaking to one at a time) while accepting a fax on 'line 2' of the SPA-2000 at the same time.

I would like to have 2 incoming (DID?) phone numbers (i.e 555-1234 and 555-1235) associated with 'line 1.'

I would like to have the option of being on a call on 'line 1' of the SPA-2000 (originating from either number) and have 'call waiting ID'
for the other incoming number during that call. (I appreciated your example of 'functionally optimized' Caller ID in your last reply)

If I do not accept the 2nd incoming call, the caller will be sent to (forwarded) to either another phone number or sent to voicemail,
based on a pre-determined time schedule configured within FreePBX.


-TriRyche