Yes I have mine working exactly like this. The following is from the Voxilla forums:
http://voxilla.com/forum-viewtopic-t-1335-postdays-0-postorder-asc-start-0.h tml The text is (in case you don't have web access). There's more posts on it but this is the nuts & guts of it. BTW, I used _ instead of A as my prefix - works great! -------- 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. Sure, the SPA-3000 does a great job of 'front-ending' inbound PSTN calls, and can even pass-through to the built-in FXS port, or external VoIP service, but I needed Asterisk to get the call BEFORE it was "answered" and handled/routed by the SPA-3000. Would seem to be a simple mode of operation, yet everywhere I looked it didn't seem possible to do just that. I wanted to use it as a 'simple' FXO <-> SIP gateway to Asterisk AND also use the FXS port as an Asterisk extension. Here's how: (I'm only detailing the tricky part .. the rest is really basic Asterisk and/or SPA-3000 setup) 1. Setup Asterisk and SPA-3000 so both the PSTN line (FXO) and Line1 (FXS) are registered with Asterisk as different extensions (i.e. FXO user ID=10 and FXS user ID=2000) on different ports (5060/5061). In this example I'll use Asterisk extension "99" as the place I want to send the inbound PSTN call to. 2. PSTN Line tab: PSTN-To-VoIP Gateway Setup PSTN-To-VoIP Gateway Enable: NO PSTN Ring Thru Line 1: YES PSTN CID For VoIP CID: YES (here's one of the tricks to make it work) PSTN CID Number Prefix: A (I used 'A' but I suppose you could pick any ALPHA character that WOULDN'T be expected as a valid caller-ID NUMBER) FXO Timer Values (sec) PSTN Ring Thru Delay: 3 3. User 1 tab: Selective Call Forward Settings Cfwd Sel1 Caller: A* Cfwd Sel1 Dest: 99 4. In Asterisk (in the context that you've defined exten 99): exten => 99,1,SETCIDNUM(${CALLERIDNUM:1}) exten => 99,2,Dial(SIP/${exten}) (for example) Here's what happens: Call rings FXO port. Wait three seconds so that caller-ID gets sent (you might need to increase this, but 3 secs seems to work fine for me) to the SPA-3000. PREFIX the caller-ID NUMBER with a LETTER before passing it to LINE 1 (so if original caller-ID was 5559991212, it's now A5559991212, not a 'valid' caller ID number, but SPA-3000 and Asterisk don't seem to care, thankfully). SELECTIVELY forward ONLY calls with caller-ID NUMBER that begin with A (actually this should be EVERY inbound PSTN call) to Asterisk extension 99 As soon as Asterisk gets the call, STRIP the 'invalid' A off and we're left with a good, original callerID number. Send the call out to a device (can be the SPA-3000 FXS (exten 2000) or port if you want!) The call is still UNANSWERED at this point. FXS port starts to ring, and original PSTN-provided caller-ID is sent as usual. Answer extension 99 (or send it voicemail) and FXO finally goes off-hook. You can make calls to extension 2000 and not worry about them being bounced back to extension 99 since no "normal" caller-ID NUMBER should ever (??) start with "A" Above all, I think this could be made a whole lot more intuitive and fool-proof if Sipura just added a feature into the firmware. > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Ed Greenberg > Sent: Wednesday, 30 March 2005 10:41 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Sipura 3000 FXO with Asterisk > > Anybody using a Sipura 3000 for FXO with Asterisk? > > Mine is working except for one small nit... > > When a call comes in from the PSTN, the Sipura answers it and > then passes it on to Asterisk, which plays extension ring tone. > > I'd prefer for the POTS line to stay on-hook while the > extension rings, and to only be answered by the Sipura when > the extension answers. > > Has anybody made this work? > > </edg> > -------------------------------------------- My mailbox is spam-free with ChoiceMail, the leader in personal and corporate anti-spam solutions. Download your free copy of ChoiceMail from www.choicemailfree.com _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
