Sipura 3000

The SPA-3000 has been discontinued by Linksys, it has been replaced by the SPA-3102 (Same features but with added router functionality).

The Sipura SPA-3000 has one FXS port and one FXO port. Both ports are fully controllable via SIP and via a local dial plan.

Sipura Webpage for SPA-3000
Linksys download webpage for SPA3000

Note: This product can be refered to as either the Sipura or Linksys : SPA3000, SPA-3000, 3000, 3k (slang)

Documentation

Sipura has excellent documentation for configuration and provisioning of their equipment.

The documentation used to be only to be available to Service Providers and Resellers.

It is now available at http://www.sipura.com/support/index.htm

Please contact sales@sipura.com if you need further information (particularly provisioning information).

There is also an unofficial page available online which has published the provisioning information, citing the reason for doing it as putting an end to Linksys' "Draconian" measures.

• Linksys (Sipura) VoIP Support Forum
• Setup Guide VOIPSpeak's Updated Guide to Using the Linksys/Sipura SPA-3000
• Setup a Linksys/Sipura SPA-3000 with FreePBX Setup instructions, troubleshooting hints, adding security, and discussion of Caller ID Name bug that prevents incoming calls when received Caller ID Name contains a single quotation mark.
• VoIP User Review
• Nice photos of the inside
• Did you know : That Sipura uses an ES3890 controller which was originally designed for playing back video CDs. Therefore analog FXO/FXS signalling requires an extra circuit from ProSLIC.
• Easy how-to on getting the SPA-3000 up running in a basic mode with VoIPStunt for the novice user.

Hotline support

added in firmware 3.1.3

From firmware release notes:
"... When a PSTN caller is automatically routed to a VoIP destination due to
    a) hotline w/o authentication, or b)call forwarding, the SPA
    will not take the FXO port off-hook until the VoIP destination answers the
    call. If the VoIP call leg fails (busy, etc), the PSTN call will not be 
    picked up by the SPA at all.
    The old behavior for this scenario is that the SPA will off-hook the FXO 
    port first before calling the VoIP destination. To keep the old behavior,
    set the new PSTN Line paramter <Off Hook While Calling VoIP> to "yes"
    (default is "no")...."

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Firmware Notes (SPA-3000 Firmware 3.1.7g)

please sign your edits! it is very antisocial to make anonymous edits. - bani

Latest Firmware is 3.1.20 (GW) (It still is being updated :D) dated 7/11/07. Download Page - dkaufman

Bugs / Feature Requests

• MAJOR: Chronic DTMF mis-detection on FXO. Voxilla Forum Thread - bani

• MINOR: Line1 erroneously re-registers and sends stun probes every second if STUN Enable: Yes and STUN Test Enable: No. Voxilla Forum Thread - bani

• MINOR: Configuration resync requests (http://blabla/admin/resync) will not acctually occur until PSTN line is plugged in. - bani

Notes / Quirks

• The echo canceller in the SPA-3000 is very basic and limited to an echo tail length of 8ms. It may take 15-30 seconds for the SPA-3000 to "converge" on the proper cancellation values for a call. Long PSTN loops or cellphone calls may suffer from uncorrectable echo with an SPA-3000, in which case the only solution is to buy an FXO gateway with a better echo canceller. - bani

• Echo Supp Enable (PSTN Line) may produce very poor quality PSTN calls, as this setting dynamically adjusts gain based on sending and receiving audio levels. Volume may jump around and callers may find their sides being squelched in a CB-halfduplex-style. If you are willing to tolerate some echo while the SPA-3000's echo canceller converges during a call, you can disable this for much better overall PSTN call quality. - bani

• In many cases the outgoing PSTN audio (SPA To PSTN Gain) is "too hot", leading to distortion that no echo canceller is able to handle. Lowering this by several db may help alleviate some PSTN echo issues. -6 is a value which works well for me. - bani

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Because people have emailed asking for my Sipura SPA-3000 config to get FXO port working with asterisk, here is what I did:

You will need the advanced screen http://192.168.1.100/admin/advanced but do not change much. From memory, I only changed the following in the "PSTN Line" tab:

Line Enable: Yes

Proxy and Registration
Proxy: astersik.domain.com

Subscriber Information
User ID: sip-username
Password: sip-password

Dial Plans
Dial Plan 2: (S0<:15551234567>)
(where 15551234567 is the extension dialed in asterisk)


PSTN-To-VoIP Gateway Setup
PSTN-To-VoIP Gateway Enable: Yes
PSTN Caller Auth Method: None
PSTN Ring Thru Line 1: no
PSTN CID For VoIP CID: yes
PSTN Caller Default DP: 2

FXO Timer Values (sec)
PSTN Answer Delay: 5

Also, look at your Disconnect tone configuration, which should meet your country specification.
For Italy, for example, it should be 425@-30,425@-30;4(.2/.2/1) where 425=tone frequency, -30dB=level, 4=number of cycles of tone 1 (third parameter in parenthesys) 0.2s ON and 0.2s OFF
Finally, check your Busy tone in Regional settings, which should meet your country specifications, i.e. for Italy should be 425@-19,425@-19;10(.5/.5/1) which mean 1 tone at 425Hz -19dB, 0.5s ON and 0.5s OFF repeated 10 times.
The tone specifications can be found at the 3amsystems.com website

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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 Adamson

See Also

• Sipura
• Geekgazette review & Article on how to make it a TRUNK LINE for Asterisk! (DEAD LINK)
• SPA Logging - Log debugging information from the Sipura to file & DB.
• Asterisk Intermittent beep with linksys SPA 3102, PAP2, DTMF sounds heard during conversation
• DTMF (dialing digits) and IVR problems in Asterisk with SPA-3102 and PAP-2 devices
• Problems dialing prefix *67 (anonymous call) on SPA-3102 and Asterisk

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I've got a way to get the SPA-3000 to use the FXO port to take inbound from PSTN (grabs and passes telco caller-ID name/num as well) and pass to Asterisk for add'l handling.

To stop the Sipura from answering incoming calls while dialing VoIP in the latest firmware there is an option "Off Hook While Calling VoIP".. When set to "No" This will mean that the call is not actually answered while VoIP is being called.. This makes it a great way to to pass off calls into the internals of Asterisk only when Asterisk is ready to answer. Further information

Where to purchase.

(List in alphabetical order please)

• 8774e4VoIP.com - Linksys SPA3000 - Linksys Certified Partner
• 888VoipStore.com - Best Prices on Linksys SPA-3000. Call for reseller pricing. 888-VOIPSTORE.
• digiumcards.com USA, International Shipping - Unbeatable Pricing!
• SoyuzSystem Specialize in Network Wireless Solution & VoIP.
• Voiplink.com SPA-3000 USA, International Shipping
• Ukraine Wildix in Ukraine

Regional Changes

Australia

For the Australian/Telstra PSTN user, the SIPURA 3000 will need to have the default hangup string changed. If you use the unit as default, you will get unpredictable hangups occuring. Using the "Silence" option can cause problems if one end of the call is "monologuing", he will get cut off so to save the embarassment, use the following hangup string.

425@-30,425@-30;1(.375/.375/1+2)

With this, you will get 4 hangup tone pulses and then the SIPURA will hangup. Enjoy

A lot of users report bad echo for caller in VOIP->PSTN (caller can hear his delayed voice during the call) and now there is a solution. Please do not decrease RX/TX gain because they really do nothing but downgrade the voice quality to the callee.

Set "FXO Port Impedance" in PSTN and Regional to 220+820||120nF. This will eliminate the callee echo (Caller hears reverberant voice from callee)

For best result, do the same thing for line 1 and set "No UDP Checksum" enabled in "SIP"

If you use Asterisk, set "canreinvite=yes" for all internal extensions. This will let RTP directly flow from Line 1 or other ATA to PSTN without going through Asterisk, thus to minimize the delay.

Try it by dialing 1800124125.

UK

On the Regional tab substitute the following:
Busy Tone: 480@-19,620@-19;10(.5/.5/1+2)
Ring Back Tone: 400@-19,450@-19;*(.4/.2/1+2,.4/2/1+2)

PSTN Line Tab:
Disconnect Tone: 400@-30;3(*/0/1)

More settings for UK: http://www.provu.co.uk/pdf/sipura/sipura_uk_regional_settings.pdf

BT has various standards which dictate exactly what the tones and technical characteristics of their lines hold.. This is VERY useful for making the Sipura sound properly like a UK line.

BT SIN 350 - Network Tones and Announcements
BT SIN 351 - Technical Characteristics Of The Single Analogue Line Interface