Getting my SPA 3000 to work

I have just rebuilt Everything… Currently running Aster 1.2.24, and trixbox 2.2.3 and Freepbx I have added a SPA 3000 (running latest 3.1.20 firmware).

I have setup a trunk to this, and a number of outbound routes - at the moment, they should all just go through to the PSTN (Different routes for Toll free numbers, international, long distance, and so on, just to make it easier to parse them all later. At the moment, this should all still be going through the PSTN (I want to get all of the “PSTN” inbound outbound stuff stable before adding any VSP’s to ensure that I can always “fall back” to known good working config…

I place a call from my local extension. (extension 25) dialling 04114XXXXX (my cell phone number (I am in Australia where cell numbers look like that…)) - My console log, does a bunch of stuff and then…

-- Executing Set("SIP/25-088a8e60", "OUTNUM=04114XXXXX") in new stack
-- Executing Set("SIP/25-088a8e60", "custom=SIP/pstn_spa") in new stack
-- Executing GotoIf("SIP/25-088a8e60", "1?gocall") in new stack
-- Goto (macro-dialout-trunk,s,22)
-- Executing GotoIf("SIP/25-088a8e60", "0?customtrunk") in new stack
-- Executing Dial("SIP/25-088a8e60", "SIP/pstn_spa/04114XXXXX|300|") in new stack
-- Called pstn_spa/04114XXXXX
-- SIP/pstn_spa-08908c40 is ringing
-- SIP/pstn_spa-08908c40 answered SIP/25-088a8e60
-- Attempting native bridge of SIP/25-088a8e60 and SIP/pstn_spa-08908c40


It rings once, appears to “answer” but then I get a PSTN dialtone, and nothing more…

What am I missing here…

I have found this page to be immensely helpful in configuring the Sipura 3000 with my system. I had installed one previously at my old house but I never needed to use the PSTN port for dialtone since I had already moved away from Bellsouth there. However, at my mom’s house, she is using Vonage. This setup page helped me get the Sipura3000 working so I can call out through her Vonage line as PSTN and fallback to my system through IAX2 if her Vonage ever goes down. Very nice. :slight_smile:

Hmm, Reset it to factory settings - applied per the aussievoip instructions, and still not getting anywhere…

The CLI does averything “right”, and then gets to the same point

– Called pstn_spa/04114xxxxx
– SIP/pstn_spa-08954530 is ringing
– SIP/pstn_spa-08954530 answered SIP/25-0891f860
– Attempting native bridge of SIP/25-0891f860 and SIP/pstn_spa-08954530

And I have external dialtone, but nothing is ringing… (or dialing…)

Dug further, found an obscure reference in the article to

try setting the Line-In-Use Voltage to a lower value, such as 15. Some telephone companies use “pair gain” or “subscriber carrier” equipment to put multiple subscribers on the same loop. If you have the great misfortune to receive telephone service through such equipment, your on-hook line voltage may be less than 30 volts, and if that is true the Sipura will think the line is always in use unless the Line-In-Use Voltage setting is lowered.

Dropped my line-is-use voltage to 15V and have a successful dial…