[tomc]

I've got a way to get the SPA3000 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 SPA3000 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 SPA3000 setup)

1. Setup Asterisk and SPA3000 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 SPA3000.

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 SPA3000 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 SPA3000 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.

[PhoneBoy]

You get points for extreme cleverness. This is definately sticky-worthy. Why not use an Underscore instead of an alpha character, though? I think that would be less likely.

[KernelSlap]

I have been working on this approach for the last 3 days, but couldn't get the last wrinkles out of the solution - the extremely clever bit was prefixing the callerid with the alpha.

Well done. My system is now working exactly as I was hoping it would.

Thanks!

[daryll]

What I did was create a dial plan that says:
(S0<:10>)
This S0 thing was listed in one of the release notes if I recall correctly.
I also set my asterisk box as my proxy
I've got the PSTN answer delay set to 12, but I want a bit of time to pick up the phone on my standard line before Asterisk picks up. You do need enough time for the caller-id to be delivered.

On the asterisk end, I defined extension 10 in the sip.conf as how the Sipura registers.

This is all under the advanced admin options for PSTN line. It works great. Asterisk picks up and gets caller-id. No additional processing is required.

Is your solution doing something different?

[ichilton]

Hi Daryll,

Yes, with your way - the SPA3000 answers the call before forwarding it on - this way doesn't.

--ian

[daryll]

Ah, I got you. When you say unanswered you really mean by unanswered by the phone company. In my solution the call is completed as soon as Sipura picks up.

With this solution, I my asterisk box could look at the caller-id or do some processing before the phone company sees the call as being connected.

Hopefully this post contributes something useful to the thread by explaining it to people as clueles as myself.

[ichilton]

Hi,

Yep - also with your setup, the caller is paying for the call while Asterisk is trying the extensions etc, with this way the call is only answered (and the caller paying) when someone or something (like voicemail) actually picks up the call.

--ian

[PhoneBoy]

In my own setup, everything either:

1. Goes straight to a VoiceMenu (therefore picking up is appropriate)
2. Goes straight to VoiceMail (i.e. I have a handset on my PSTN line and either pick it up or don't before Asterisk picks up)

Either way, this is still clever.

[enzo86]

as soon as i turn on:
PSTN CID For VoIP CID: YES i get:
chan_sip.c:7208 handle_request: Failed to authenticate user WIRELESS CALLER <sip:A9193<removed>@


apollo*CLI> sip show peers
Name/username Host Dyn Nat ACL Mask Port Status
3100/3100 192.168.1.12 D 255.255.255.255 5060 OK (72 ms)
3000/3000 192.168.1.12 D 255.255.255.255 5060 OK (114 ms)
spa3k/asterisk 192.168.1.14 255.255.255.255 5061 OK (22 ms)
2003/2003 192.168.1.14 D 255.255.255.255 5061 Unmonitored
2002/2002 192.168.1.14 D 255.255.255.255 5060 OK (17 ms)
fwd/52766 69.90.155.70 255.255.255.255 5060 OK (47 ms)
fwd-sip/428114 69.90.155.70 255.255.255.255 5060 OK (52 ms)
7 sip peers loaded [7 online , 0 offline]
apollo*CLI> sip debug peer 2003
SIP Debugging Enabled for IP: 192.168.1.14:5061
apollo*CLI>

Sip read:
INVITE sip:2003@192.168.1.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.14:5061;branch=z9hG4bK-89b39f60
From: WIRELESS CALLER <sip:A919349REMOVED@192.168.1.10:5060>;tag=387c72a 723db44c8o1
To: <sip:2003@192.168.1.10:5060>
Call-ID: 7fd00353-c9670c0c@192.168.1.14
CSeq: 101 INVITE
Max-Forwards: 70
Contact: NO CALLER-ID <sip:A919REMOVED@192.168.1.14:5061>
Expires: 240
Diversion: sipura <sip:2002@192.168.1.10:5060>;reason=user-busy
User-Agent: Sipura/SPA3000-2.0.11(GWg)
Content-Length: 233
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 507076 507076 IN IP4 192.168.1.14
s=-
c=IN IP4 192.168.1.14
t=0 0
m=audio 16434 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

15 headers, 12 lines
Using latest request as basis request
Sending to 192.168.1.14 : 5061 (non-NAT)
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.14:16434
Found description format PCMU
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x106(GSM|ULAW|G729A), peer - audio=0x4(ULAW)/video=0x0(EMPTY), combined - 0x4(ULAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723)
Found peer 'spa3k'
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.14:5061;branch=z9hG4bK-89b39f60
From: WIRELESS CALLER <sip:A9193REMOVED7@192.168.1.10:5060>;tag=387c72a7 23db44c8o1
To: <sip:2003@192.168.1.10:5060>;tag=as513a35c9
Call-ID: 7fd00353-c9670c0c@192.168.1.14
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2003@192.168.1.10>
Proxy-Authenticate: Digest realm="asterisk", nonce="27b671b3"
Content-Length: 0


to 192.168.1.14:5061
Scheduling destruction of call '7fd00353-c9670c0c@192.168.1.14' in 15000 ms
apollo*CLI>

Sip read:
ACK sip:2003@192.168.1.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.14:5061;branch=z9hG4bK-89b39f60
From: WIRELESS CALLER <sip:A9193REMOVED@192.168.1.10:5060>;tag=387c72a72 3db44c8o1
To: <sip:2003@192.168.1.10:5060>;tag=as513a35c9
Call-ID: 7fd00353-c9670c0c@192.168.1.14
CSeq: 101 ACK
Max-Forwards: 70
Contact: NO CALLER-ID <sip:A919REMOVED@192.168.1.14:5061>
User-Agent: Sipura/SPA3000-2.0.11(GWg)
Content-Length: 0


10 headers, 0 lines
apollo*CLI>

Sip read:
INVITE sip:2003@192.168.1.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.14:5061;branch=z9hG4bK-5386f724
From: WIRELESS CALLER <sip:A91934REMOVED@192.168.1.10:5060>;tag=387c72a7 23db44c8o1
To: <sip:2003@192.168.1.10:5060>
Call-ID: 7fd00353-c9670c0c@192.168.1.14
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest username="2003",realm="asterisk",nonce="27b671b3", uri="sip:2003@192.168.1.10:5060",algorithm=MD5,res ponse="7eafe51e22b9c7226cb3e9db017ec045"
Contact: NO CALLER-ID <sip:A9193499847@192.168.1.14:5061>
Expires: 240
Diversion: sipura <sip:2002@192.168.1.10:5060>;reason=user-busy
User-Agent: Sipura/SPA3000-2.0.11(GWg)
Content-Length: 233
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 507076 507076 IN IP4 192.168.1.14
s=-
c=IN IP4 192.168.1.14
t=0 0
m=audio 16434 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

16 headers, 12 lines
Using latest request as basis request
Sending to 192.168.1.14 : 5061 (non-NAT)
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.14:16434
Found description format PCMU
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x106(GSM|ULAW|G729A), peer - audio=0x4(ULAW)/video=0x0(EMPTY), combined - 0x4(ULAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723)
Found peer 'spa3k'
Nov 27 02:56:56 NOTICE[8041]: chan_sip.c:7208 handle_request: Failed to authenticate user WIRELESS CALLER <sip:A9193REMOVED@192.168.1.10:5060>;tag=387c72a72 3db44c8o1
Reliably Transmitting (no NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.1.14:5061;branch=z9hG4bK-5386f724
From: WIRELESS CALLER <sip:A9193REMOVED@192.168.1.10:5060>;tag=387c72a72 3db44c8o1
To: <sip:2003@192.168.1.10:5060>;tag=as513a35c9
Call-ID: 7fd00353-c9670c0c@192.168.1.14
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2003@192.168.1.10>
Content-Length: 0


to 192.168.1.14:5061
apollo*CLI>

Sip read:
ACK sip:2003@192.168.1.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.14:5061;branch=z9hG4bK-5386f724
From: WIRELESS CALLER <sip:A9193REMOVED@192.168.1.10:5060>;tag=387c72a72 3db44c8o1
To: <sip:2003@192.168.1.10:5060>;tag=as513a35c9
Call-ID: 7fd00353-c9670c0c@192.168.1.14
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username="2003",realm="asterisk",nonce="27b671b3", uri="sip:2003@192.168.1.10:5060",algorithm=MD5,res ponse="6c5ca22046cf5389c4f3855fbd4bd58e"
Contact: NO CALLER-ID <sip:A9193499847@192.168.1.14:5061>
User-Agent: Sipura/SPA3000-2.0.11(GWg)
Content-Length: 0


11 headers, 0 lines
Destroying call '7fd00353-c9670c0c@192.168.1.14'
apollo*CLI> sip no debug
SIP Debugging Disabled

Originally Posted by tomc

I've got a way to get the SPA3000 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.

[cut]

3. User 1 tab:

Selective Call Forward Settings

Cfwd Sel1 Caller: A*
Cfwd Sel1 Dest: 99

[cut]

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 SPA3000.

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 SPA3000 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

[cut]

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.

I just got my SPA3000 and tried this out with my Verizon PSTN. I basically used the Voxilla spaasterisk configurator and then made the above tweaks. The only problem I have is that if I disable the callerid in my cell phone (T-mobile) and call my number it rings line 1 instead of forwarding to Asterisk.

I changed the Selective forward from A* to * as a test and the SPA3000 is sending the username as the CID in that case. I changed the username to Asippstn on the off chance that would work and it did not. I think I need a way to match both A* and an empty CID and then tweak my asterisk config blocked CID calls to Line 1 get a different CID.

On the whole this would be much easier if Sipura added an option to do this automatically.