Asterisk Forum

http://www.sipura.com/support/spa3000faq/Section_3.html#4

Hi,

John Covert wrote: I found another way to connect my SPA-3000 to Asterisk.  The advantage is that the PSTN line is not answered until Asterisk actually answers the call.

Has anyone else had this working?

Thanks

--ian

Hi,

Also, John - if you are still around, is there any chance you could post or send over the relevent parts of your asterisk and spa configs?

Thanks

--ian

John's instructions do the trick (nice find), but I'll add that I find that caller-ID isn't being passed all the way to the final SIP extension :(

I end up seeing the PSTN Line 'user id' (the same as is in <Cfwd Sel1 Caller>) instead of the telco-provided caller-id info. I have <PSTN CID For VoIP CID> = Yes and have tried various time settings for <PSTN Ring Thru Delay>.

Might be something I'm doing wrong, of course and I hope, but I tend to think that this seemingly simplistic feature (call it 'simple one-stage PSTN-VoIP gateway with caller-ID passthru') is somehow just not possible given the current firmware. I see lots of references in the manual to "VoIP gateway auto-answers the call ..."

If someone figures out how to do this I'll be 100% satisfied (I'm already very impressed with the product) as it will suit my needs perfectly.

FYI - firmware is version 2.0.11(GWa)

I'll post some details later today when I'm back home. That should make it crystal clear.

My next step is to bring this up on voxilla, but, alas, have been busy doing 'real' work for the past few days. Now that I have my second SPA-3000, I'm _really_ interested in getting past this hurdle. Hopefully it's something Sipura will address or clarify.

Unfortuately the few specific messages I've come across so far seem to flatly state, "it's not possible" (even phoneboy said that I think), "not by design," "resource constrained" ... blah blah. I, optimistically, don't believe it but haven't come across an "offical" response from Sipura one way or the other. My hunch is they could do it, rather easily, since it's 99% of the way there (see below).

Through further experimentation I AM able to make caller-ID work, but then came across another stumbling block for the overall plan.

The caller-ID info CAN and WILL be passed to the final SIP destination, before "answering" the PSTN line IF:

1. Caller-ID was supplied on the original call (of course)

2. <PSTN CID For VoIP CID> = Yes

3. <PSTN-To-VoIP Gateway Enable> = No

Here's where it gets tricky:

If, in the User 1 tab, <Cfwd Sel1 Caller> set to the <user id> of the PSTN line, and <Cfwd Sel1 Dest> set to an extension in the asterisk context for the PSTN line ... works ONLY when there's no caller-ID supplied/received or passed-on from PSTN.

The selective forwarding works at this point because it matches the <user ID> of the PSTN line, but the caller-ID passed to the SIP extension is the PSTN line's <user ID> in this case.

To work past that, make sure that <PSTN Ring Thru Delay> is set high enough (i.e. 3 or 4 secs in US) to allow for complete reception of FSK transmission from telco, before ringing thru AND ... here's the kicker ... you CAN'T selective forward on the <user ID> anymore because it's now the CALLER-ID!!!!

So you have to call-forward "always" to make it work now. You can do things like call-forward-selective on, say, 516*, which might help out some folks.

I haven't done anything past this right now, but maybe there's something more that can be done with the wildcard matching in the call-forward fields and tricky digit strip/replace dial-plans or some such ... time will tell.

Keen observer will then say ... but what if I want to use <Line 1> for "regular" SIP/FXS functions. That's the rub ... you CAN'T since it'll always be forwarded.

So close, yet so far.

1. Start with 'factory defaults'
2. Assign different "User ID:" to Line 1 and PSTN Line and make sure they register with asterisk (set IP of asterisk server in Proxy: on Line 1 and PSTN Line tabs).

PSTN Line - PSTN-To-VoIP Gateway Setup 

PSTN-To-VoIP Gateway Enable: no
PSTN Ring Thru Line 1: yes
PSTN CID For VoIP CID: yes   

PSTN Line - FXO Timer Values (sec)

PSTN Ring Thru Delay: 3
 

User 1 - Selective Call Forward Settings 

Cfwd Sel1 Caller: *  Cfwd Sel1 Dest:  11

(where 11 is the extension defined on my asterisk server that I want to ring when call comes in via PSTN).