Linksys spa3102 on 3CX v12 can not make calls in or out

[Jonners59]

I am in a flap
I have had my 3CX working wonderfully for a number of years with a Grandstream FXO: GWX4104. The Gradstream failed and I needed a new device, but with only two PSTN lines I thought that another GWX was a bit over kill. I subsequently bought 2 x linksys spa3102 devices, one for each and have upgraded my 3CX from version 9 to 12...

I just can't make this work. In or outbound. The two Lynxisys devices work when the analogue phone is connected directly, but not as an FXO. I have followed everything I have found on line. Can anyone help, please. Please see below for sys log for an outbound call. The addition of a 4 is to force an outbound call via the FXO instead of the SIP gateway/service provider. This is stripped

Inbound the call does not even show in the logs... and yes the devices are registered.

23-Nov-2013 11:07:56.625   L:47.1[Extn] got Terminated Recv Req CANCEL from 192.168.1.90:5060 tid=798c9140c2c196ab1ca32280f60eff11 Call-ID=704268022@192_168_1_90:
CANCEL sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK798c9140c2c196ab1ca32280f60eff11;rport=5060
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected];user=phone>
From: "Baroni Limited"<sip:[email protected]>;tag=2294670366
Call-ID: 704268022@192_168_1_90
CSeq: 3 CANCEL
Proxy-Authorization: Digest username="200", realm="3CXPhoneSystem", algorithm=MD5, uri="sip:[email protected];user=phone", nonce="414d535c08a11d6717:9207c2074ab942b3fd076ac4f35e9abc", response="cf81298bb9a8ef4fdd170651f4bdd233"
User-Agent: N300 IP/42.075.00.000.000
Content-Length: 0
23-Nov-2013 11:07:56.625   Terminated from "Baroni Limited"<sip:[email protected]>;tag=2294670366 to <sip:[email protected];user=phone>;tag=fe23ec71; reason: RemoteBye
23-Nov-2013 11:07:56.625   L:47.1[Extn] Sending: OnSendResp Send 487/INVITE from 0.0.0.0:0 tid=798c9140c2c196ab1ca32280f60eff11 Call-ID=704268022@192_168_1_90:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK798c9140c2c196ab1ca32280f60eff11;rport=5060
To: <sip:[email protected];user=phone>;tag=fe23ec71
From: "Baroni Limited"<sip:[email protected]>;tag=2294670366
Call-ID: 704268022@192_168_1_90
CSeq: 3 INVITE
Content-Length: 0
23-Nov-2013 11:07:56.625   SendMsg from <sip:[email protected];user=phone>;tag=fe23ec71 to "Baroni Limited"<sip:[email protected]>;tag=2294670366
23-Nov-2013 11:07:56.625   L:47.1[Extn] Sending: OnSendResp Send 200/CANCEL from 0.0.0.0:0 tid=798c9140c2c196ab1ca32280f60eff11 Call-ID=704268022@192_168_1_90:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK798c9140c2c196ab1ca32280f60eff11;rport=5060
Contact: <sip:[email protected];user=phone>
To: <sip:[email protected];user=phone>;tag=fe23ec71
From: "Baroni Limited"<sip:[email protected]>;tag=2294670366
Call-ID: 704268022@192_168_1_90
CSeq: 3 CANCEL
Content-Length: 0
23-Nov-2013 11:07:56.625   SendMsg from <sip:[email protected];user=phone>;tag=fe23ec71 to "Baroni Limited"<sip:[email protected]>;tag=2294670366
23-Nov-2013 11:07:52.345   L:47.2[Line:10001>>02086689100]: Terminating targets, reason: SIP ;cause=408 ;text="Request Timeout"
23-Nov-2013 11:07:52.345   Leg L:47.2[Line:10001>>02086689100] is terminated: Cause: 408 Request Timeout/INVITE from local
23-Nov-2013 11:07:52.345   L:47.2[Line:10001>>02086689100] got Terminated Recv 408/INVITE from 0.0.0.0:0 tid=470ece4f0b13026f Call-ID=MDIyN2RjZjc5MTRkNmNjZDM5MTQ5YjZlMjg1MTlhZTQ.:
SIP/2.0 408 Request Timeout
Via: SIP/2.0/UDP 192.168.1.51:5060;branch=z9hG4bK-d8754z-470ece4f0b13026f-1---d8754z-;rport
To: <sip:[email protected]:5062>;tag=17486b5c
From: "Baroni Limited"<sip:[email protected]:5060>;tag=dd4c2032
Call-ID: MDIyN2RjZjc5MTRkNmNjZDM5MTQ5YjZlMjg1MTlhZTQ.
CSeq: 1 INVITE
Content-Length: 0
23-Nov-2013 11:07:52.345   Terminated from <sip:[email protected]:5062>;tag=17486b5c to "Baroni Limited"<sip:[email protected]:5060>;tag=dd4c2032; reason: Error
23-Nov-2013 11:07:52.345   ~Target=PSTNline:02086689100@(Ln.10001@OFFICE)
23-Nov-2013 11:07:52.345   [CM503025]: Call(C:47): Calling T:Line:10005>>02086689100@[Dev:sip:[email protected]:5060] for L:47.1[Extn]
23-Nov-2013 11:07:52.344   Route to L:47.3[Line:10005>>02086689100] sends Invite-OUT Send Req INVITE from 0.0.0.0:0 tid=c96f1d2adc69cd6a Call-ID=ZTkyMTU5MmQ4NjJlZjQ4NDZjMWNjOTIxMTliYjdlYmI.:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/ ;branch=z9hG4bK-d8754z-c96f1d2adc69cd6a-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]:5060>
From: "Baroni Limited"<sip:[email protected]:5060>;tag=ed69824b
Call-ID: ZTkyMTU5MmQ4NjJlZjQ4NDZjMWNjOTIxMTliYjdlYmI.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
Content-Length: 279

v=0
o=3cxPS 5351931904 123346092033 IN IP4 82.14.176.51
s=3cxPS Audio call
c=IN IP4 82.14.176.51
t=0 0
m=audio 9044 RTP/AVP 0 8 3 99 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:99 SPEEX/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
23-Nov-2013 11:07:52.344   Dev(1):[sip:[email protected]:5060 / 844231085]: PBX contact is resolved to public IP: <sip:[email protected]:5060>
23-Nov-2013 11:07:52.171   Outbound URI is used: sip:77.240.48.201:5065
23-Nov-2013 11:07:52.167   Added leg L:47.3[Line:10005>>02086689100]
23-Nov-2013 11:07:52.167   ~Route=Dev:sip:[email protected]:5060
23-Nov-2013 11:07:52.167   Call to T:Line:10001>>02086689100@[Dev:sip:[email protected]:5060] from L:47.1[Extn] failed, cause: Cause: 408 Request Timeout/INVITE from local
23-Nov-2013 11:07:52.163   L:47.2[Line:10001>>02086689100] got Failure: Failure Recv 408/INVITE from 0.0.0.0:0 tid=470ece4f0b13026f Call-ID=MDIyN2RjZjc5MTRkNmNjZDM5MTQ5YjZlMjg1MTlhZTQ.:
SIP/2.0 408 Request Timeout
Via: SIP/2.0/UDP 192.168.1.51:5060;branch=z9hG4bK-d8754z-470ece4f0b13026f-1---d8754z-;rport
To: <sip:[email protected]:5062>;tag=17486b5c
From: "Baroni Limited"<sip:[email protected]:5060>;tag=dd4c2032
Call-ID: MDIyN2RjZjc5MTRkNmNjZDM5MTQ5YjZlMjg1MTlhZTQ.
CSeq: 1 INVITE
Content-Length: 0
23-Nov-2013 11:07:52.157   [CM503003]: Call(C:47): Call to <sip:[email protected]:5062> has failed; Cause: 408 Request Timeout/INVITE from local
23-Nov-2013 11:07:52.155   Session 95886 has failed in leg L:47.2[Line:10001>>02086689100] ; Cause: 408 Request Timeout/INVITE from local
23-Nov-2013 11:07:52.150   Failure from <sip:[email protected]:5062>;tag=17486b5c to "Baroni Limited"<sip:[email protected]:5060>;tag=dd4c2032
23-Nov-2013 11:07:40.886   Currently active calls - 1: [47]
23-Nov-2013 11:07:20.022   [CM503025]: Call(C:47): Calling T:Line:10001>>02086689100@[Dev:sip:[email protected]:5060] for L:47.1[Extn]
23-Nov-2013 11:07:20.022   Route to L:47.2[Line:10001>>02086689100] sends Invite-OUT Send Req INVITE from 0.0.0.0:0 tid=7609461399746f36 Call-ID=MDIyN2RjZjc5MTRkNmNjZDM5MTQ5YjZlMjg1MTlhZTQ.:
INVITE sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/ ;branch=z9hG4bK-d8754z-7609461399746f36-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]:5062>
From: "Baroni Limited"<sip:[email protected]:5060>;tag=dd4c2032
Call-ID: MDIyN2RjZjc5MTRkNmNjZDM5MTQ5YjZlMjg1MTlhZTQ.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
Content-Length: 357

v=0
o=3cxPS 441525993472 356364845057 IN IP4 192.168.1.51
s=3cxPS Audio call
c=IN IP4 192.168.1.51
t=0 0
m=audio 7170 RTP/AVP 0 8 3 13 9 110 99 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:13 CN/8000
a=rtpmap:9 G722/8000
a=rtpmap:110 iLBC/8000
a=rtpmap:99 SPEEX/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
23-Nov-2013 11:07:19.996   Outbound URI is used: sip:[email protected]:5060
23-Nov-2013 11:07:19.983   Added leg L:47.2[Line:10001>>02086689100]
23-Nov-2013 11:07:19.927   [Flow] Call(C:47): making call from L:47.1[Extn] to T:Line:10002>>02086689100@[Dev:sip:[email protected]:5060]
23-Nov-2013 11:07:19.927   [CM503027]: Call(C:47): From: Extn:200 ("Baroni Limited" <sip:[email protected]:5060>)  to  T:Line:10002>>02086689100@[Dev:sip:[email protected]:5060]
23-Nov-2013 11:07:19.927   [CM503004]: Call(C:47): Route 3: from L:47.1[Extn] to T:Line:10002>>02086689100@[Dev:sip:[email protected]:5060]
23-Nov-2013 11:07:19.927   Line limit check: Current # of calls for line Lc:10002(@HOME[<sip:[email protected]:5060>]) is 0; limit is 1
23-Nov-2013 11:07:19.927   [Flow] Call(C:47): making call from L:47.1[Extn] to T:Line:10005>>02086689100@[Dev:sip:[email protected]:5060]
23-Nov-2013 11:07:19.927   [CM503027]: Call(C:47): From: Extn:200 ("Baroni Limited" <sip:[email protected]:5060>)  to  T:Line:10005>>02086689100@[Dev:sip:[email protected]:5060]
23-Nov-2013 11:07:19.927   [CM503004]: Call(C:47): Route 2: from L:47.1[Extn] to T:Line:10005>>02086689100@[Dev:sip:[email protected]:5060]
23-Nov-2013 11:07:19.927   Line limit check: Current # of calls for line Lc:10005(@VoIPTalk[<sip:[email protected]:5060>]) is 0; limit is 2
23-Nov-2013 11:07:19.927   [Flow] Call(C:47): making call from L:47.1[Extn] to T:Line:10001>>02086689100@[Dev:sip:[email protected]:5060]
23-Nov-2013 11:07:19.927   [CM503027]: Call(C:47): From: Extn:200 ("Baroni Limited" <sip:[email protected]:5060>)  to  T:Line:10001>>02086689100@[Dev:sip:[email protected]:5060]
23-Nov-2013 11:07:19.927   [CM503004]: Call(C:47): Route 1: from L:47.1[Extn] to T:Line:10001>>02086689100@[Dev:sip:[email protected]:5060]
23-Nov-2013 11:07:19.927   Line limit check: Current # of calls for line Lc:10001(@OFFICE[<sip:[email protected]:5060>]) is 0; limit is 1
23-Nov-2013 11:07:19.927   Call(C:47): Call from Extn:200 to 402086689100 matches outbound rule 'PSTN Baroni'
23-Nov-2013 11:07:19.927   [Flow] Call(C:47): has built target endpoint: Out#:>>Rule{PSTN Baroni}>>402086689100 for call from L:47.1[Extn]
23-Nov-2013 11:07:19.927   [Flow] Target endpoint for 402086689100 is Out#:>>Rule{PSTN Baroni}>>402086689100
23-Nov-2013 11:07:19.927   Selected prefix: 4
23-Nov-2013 11:07:19.927   Looking for outbound rule: dialed = [402086689100], processed: [402086689100]; from-ext:
23-Nov-2013 11:07:19.927   [Flow] Building target endpoint to 402086689100 from "Baroni Limited" <sip:[email protected]:5060>
23-Nov-2013 11:07:19.927   [CM503010]: Call(C:47): Making route(s) from Extn:200 to <sip:[email protected]:5060>
23-Nov-2013 11:07:19.927   Remote SDP is set for leg L:47.1[Extn]
23-Nov-2013 11:07:19.927   OnOffer from "Baroni Limited"<sip:[email protected]>;tag=2294670366
23-Nov-2013 11:07:19.926   [CM505001]: Endpoint Extn:200: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [N300 IP/42.075.00.000.000] PBX contact: [sip:[email protected]:5060]
23-Nov-2013 11:07:19.905   Inbound DID: ''; Phonebook Name: ''
23-Nov-2013 11:07:19.905   [CM500002]: Call(C:47): Info on incoming INVITE from Extn:200:
Invite-IN Recv Req INVITE from 192.168.1.90:5060 tid=798c9140c2c196ab1ca32280f60eff11 Call-ID=704268022@192_168_1_90:
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK798c9140c2c196ab1ca32280f60eff11;rport=5060
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected];user=phone>
From: "Baroni Limited"<sip:[email protected]>;tag=2294670366
Call-ID: 704268022@192_168_1_90
CSeq: 3 INVITE
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Proxy-Authorization: Digest username="200",realm="3CXPhoneSystem",algorithm=MD5,uri="sip:[email protected];user=phone",nonce="414d535c08a11d6717:9207c2074ab942b3fd076ac4f35e9abc",response="371787970a59a02275636051f2acc0d6"
Supported: replaces
User-Agent: N300 IP/42.075.00.000.000
Allow-Events: message-summary, refer, ua-profile, talk
Content-Length: 381

v=0
o=200 5006 197 IN IP4 192.168.1.90
s=Mapping
c=IN IP4 192.168.1.90
t=0 0
m=audio 5006 RTP/AVP 9 8 0 96 97 2 18 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:97 AAL2-G726-32/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
23-Nov-2013 11:07:19.905   [CM503001]: Call(C:47): Incoming call from Extn:200 to <sip:[email protected]:5060>
23-Nov-2013 11:07:19.868   Outbound URI is used: sip:[email protected]:5060
23-Nov-2013 11:07:19.868   IncomingCall: C:47 from <sip:[email protected]:5060> to <sip:[email protected]:5060>
23-Nov-2013 11:07:19.868   Added leg L:C:47.1[No endpoint yet]
23-Nov-2013 11:07:19.865   UasSession 95857 started
23-Nov-2013 11:07:19.865   Call from "Baroni Limited"<sip:[email protected]>;tag=2294670366 to <sip:[email protected];user=phone>;tag=fe23ec71
23-Nov-2013 11:07:10.769   Currently active calls [none]

 

[Jonners59]

I am now getting very frustrated with this. Should be a simple set-up, but this box is far too complex.

I have got the standard dialling from the PSTN /FXO to the 3CX working fine, and ordinarily outbound calls via the gateway are not too important as outbound calls route via my VoIP SP, but I want them enabled to allow my wife to use the phone for some of her own bits n bobs...

So to have 3CX to PSTN/FXO and Handset/FXS to PSTN/FXO working is also important, whilst less so.

I did have these working, but then they stopped, but I do not know why and what has changed, I think it was when I went from the default FW to the latest FW. I have tried by passing the gateway and connecting my handset directly in to the BT socket and that works. I have also tried turning the gateway off (failover) and that also works, but when it is on; I get dial tone, can hear the numbers dialling, but the call fails with a tone. As if BT is just not recognising what I am dialling. The same also happens on my Virgin Media line. Needless to say, calls out from the 3CX also fail. My settings are no different to those I have been sent or read, although everyone has slight differences, no combination seems to work.

The attached screen shot shows what the gateway is seeing - clearly the number is being dialled, note "Call 1 Peer Phone: This clearly shows the dialled number, however.
Call 1 State: says "Invalid", so something went wrong. If I can fix this then I think it all should work perfectly.

I had read that the latest 5... FW version does not work in the UK, so I downgraded to the one recommended 3.6...... And still it does not work.

Some of the resources used:
http://blog.trumpton.org.uk/2010/10/linksys-spa3102.html
http://blog.trumpton.org.uk/2010/10/uk-dial-and-progress-tones.html
http://www.3amsystems.com/wireline/tone-search.htm
http://www.solidfluid.co.uk/sfsite.php/00000222
http://www.aoakley.com/articles/2008-01-08.php
http://www.sinet.bt.com/350v1p3.pdf
http://www.cisco.com/en/US/products/ps10027/index.html
FW Updates
http://software.cisco.com/download/type.html?mdfid=282414112&catid=null

 

[leejor]

Did you do a "factory" reset before changing settings again? There may have been "old" changes, you did long ago that are affecting things. Yes, the 3102 is VERY complicated, because it can do a lot, but, for operation with 3CX (in most cases) there is not a lot that is required to be changed. Being outside North America does, however, warrant some additional changes from default settings.

The screen-shot you posted shows nothing other than LINE 1 is registered (so you decided to register it with 3CX after all?) , and off Hook along with the last attempted call on GW0.

You will have to post additional "Tabs", both the full PSTN, and Line 1, to start with, before I'd have any idea what you have "going on" in the 3102.

 

[Jonners59]

Hi Leejor

Did you do a "factory" reset before changing settings again? There may have been "old" changes, you did long ago that are affecting things.

Yes, full rest - took an age to get back in to the beastie, but once in I followed my steps backwards again, knowing I knew how to get back to where I left off if all else failed.

Yes, the 3102 is VERY complicated, because it can do a lot, but, for operation with 3CX (in most cases) there is not a lot that is required to be changed. Being outside North America does, however, warrant some additional changes from default settings.

There is FAR more outside the US to change and to be honest all of it could be pre packed, it doesn't change and would ensure a speedier and safer install. We should be able to select our country and the regional and some other bits pre-populate, like other makes have taken the trouble to do. And yes, it does a lot, but so do others... No, I think the Cisco/Linksys boys and girls could make it much easier. But it is what it is.

The screen-shot you posted shows nothing other than LINE 1 is registered (so you decided to register it with 3CX after all?) , and off Hook along with the last attempted call on GW0.

Well I have had that all along, but due to complexity I am not sure what it means and why it has to be there. I have, as you know, two accounts set up on the 3CX for the gateway(s), I just register to those. The inbound works well. The outbound fails, but I think that is caused by the same problem that the outbound from the FXS is having. This is a screenshot during an attempted outbound call from the FXO to a mobile phone 07770......, so yes off hook and yes out via GW0.

You will have to post additional "Tabs", both the full PSTN, and Line 1, to start with, before I'd have any idea what you have "going on" in the 3102.

OK, coming over... Can I send via private message or eMail, please?

PS: Tried to send you a PM. Not sure you got it as it sits in my Outbox and refuses to move.

 

[leejor]

PS: Tried to send you a PM. Not sure you got it as it sits in my Outbox and refuses to move.

Because it's the weekend and I have other things to do than sit at a computer...

Had a quick look over your configuration. As I get the time (tomorrow) I may will find addition issues.
The first ones that stand out.. You have line one set with the Same port as the PSTN line and registered using the same user/password as the trunk, you can't do that. The line has to use a different port, usually 5060, and must be registered as a 3CX extension, not a trunk. That is why your calls are going back into 3CX. You may try leaving the line enabled but with no proxy info, but say yes to Make Call without Reg. It may or may not work for you. You may just have to register it as an extension and hope the dialplan sends all calls out on the PSTN port.

Under the PSTN tab, check the syntax of Dial plan 8, it is not correct. Re-read this link...http:/ /www.3cx.com/voip-gateways/linksys-spa3102/ - NO LONGER AVAILABLE


I'm not certain that the Caller ID FSK Standard should be Bell 202, if caller id still isn't working, that might be it.

After going over your settings again, nothing else really stood out as an issue, but, that's not to say that i didn't miss something.. You may have to enable the LINE 1, and register it as a 3CX extension to be able to place any calls over it at all. I'm not sure if it will work when disabled, I've never tried that. I'm still thinking that everything considered, you'd be better off just plugging the phone directly into a spare PSTN socket. using the FXS socket when you only want to place calls out on the PSTN anyway, doesn't make sense. later, if you want to be able to route some calls to PSTN and some to your VoIP provider you can use the FXS socket and create a Dialplan in the 3102 to handle the dialled digits.

 

[Jonners59]

Leejoy
You solved it.. It was JUST that I had to change the port number!!!!! Does EXACTLY what I wanted. THANK YOU :lol:

Jonners

 

[leejor]

Did you disable Line 1, or change it to register as a 3CX extension, as I suggested?

By using a combination of devices routing through 3CX and connected directly to the PSTN line, then saying some calls work correctly, it's hard to know which ones you are talking about.

 

[Jonners59]

I just changed the port number in Line 1 to 5060, as opposed to the assigned port number 5062 which I use in the PSTN page... THAT is ALL that was needed. So silly. I knew it must be doable as I had it working before I tried to fix the other issues and changed all the settings.

I can now make calls out over the PSTN automatically if the VoIP route fails (alternative route) or I can automatically force it with the routing table on the 3CX for specific calls that need to go out over PSTN, such as the emergency services (999) and I can also force it manually by adding a 4 or 5 (depending on whether Home or Business line) out over the PSTN from the 3CX so I now have a full fall back.

I can also select the fixed line on the handset and make a direct call over the PSTN, via the ATA, which is not too important but useful if the 3CX fails, and whilst doing so I do not have the handset ringing when inbound calls come in via the PSTN line, but via the 3CX thus the 3CX is in control... It only kicks in if the 3CX fails. It's perfect, simple and highly flexible and automated.

The code below is the VOICE setup for UK Land line taking the official OFCOM and UK standards and my 3CX settings. Also set that All calls of all types are routed out from the 3CX to the PSTN port... I am assuming you are using the 3CX to do the routing tables, if of any use to anyone else.

Attached is my html configuration file zipped... It should work. My systems personal data has been removed, so account details and server details will be needed. Note that the Server SIP port is 5060 and the post for the gateway device on the 3CX has to be set to 5062 all standard stuff, but worth remembering.