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

[Cjay]

I've just upgraded both my 3102's - the latest firmware seems to be 5.2.13(GW002).

There is one trick that you can use to fully restore a previously saved SPA3102 configuration (rather than having to go through a printout and manually change all the parameters). Once you have a working config then save both the the router & voice configs to .htm files (as discussed above).

Edit both these files (in notepad) to replace the only occurrence of the line:
<form action="asipura.spa" method="POST">
with:
<form action="http://IP/admin/asipura.spa" method="POST">
[where IP is address of your SPA-3102]

And then resave as .htm again.

To restore the SPA3102 from the saved configuration, open the saved/edited file in a browser and hit "Submit All Changes" (you need to do this for each of the router & voice backup files).

Can save a lot of time if for example you want to quickly reconfigure a shelf-spare.

Chris

 

[Jonners59]

Thanks Leejor

Ok...

Caller ID, You have to have the CID setting in the 3102 match what is being send by your provider. If it is BT, then I assume that you were sent some set-up information by another UK user that shows what settings are required. If you believe that the 3102 is not passing on the correct information, then check the Voice/Info tab, look for Last PSTN caller. If the 3102 got the number, it will show up there. As for name display, I think you are out of luck as I'm not aware of it being sent in the UK, unless by some cable companies. The 3102 will pass he name you have assigned. You can use the Phone Book in 3CX to substitute a name of your choice per number match.

I am using the settings sent. The FXO is showing the details of the last caller in the status. It is not showing the details at the 3CX end, yet in the activity log the details are showing

Contact: <sip:[email protected]:5060>

* note number changed by me fr security

Echo...Under the PSTN tab at the bottom, you will see SPA to PSTN gain. and PSTN to SPA gain. This is where you adjust the levels. The default is zero in each case. If you are hearing echo, it may mean that the level that you are sending is to "hot". There are other settings that can affect level besides the level adjustments. The FXO port impedance can introduce echo if not set correctly and also under the PSTN tab/Audio there are additional echo setting that you can "play" with. Keep note of what you change.

That sounds like a lot of guess work and trial and error. I'll see first if anyone can suggest some settings to try.

Failure...using the 3102 with as a 3CX trunk is not really the use that the failure option was intended. It is simply a power failure option designed to allow dialtone on the phone attached to the FXS port when used in a situation where the Line dialplan in the 3102 routes all calls to both the PSTN port and a VoIP provider. If 3CX does not see the gateway registered, the only alternative is the next trunk group in the outbound rules. If that isn't happening for you, then check the 3CX logs to find out why. In your case I would avoid plugging a phone into the FXS port and just use your 3CX extensions to place calls.

OK, that's interesting and noted. Seems a bit of a shortfall of the device to me. I would have expected the techies could have wporked out a simple no Ethernet/no registration go to call fail-over.... The nasty Chinese device I tried for all it's failings was able to do this. Now I know I nee to decide if it is better fto just plug the DECT phones straight in to the socket in the wall or the gateway, the res=ult seems to be the same, but at least the former i have a fail-over if Ethernet or registration fail, the down side is the DECT rings when I really only want to rely on the 3CX.

As far as your 3CX dialplans go...I'm not sure why your outbound rules have those digits in the "calls to Numbers with a length of" field.
Your outside numbers must be longer than that. If you are dialling a number such as 13105551212, then the length is 11 digits.
If you are dialling the emergency number (999), then the digit length is 3
If you are dialling a London number (02075461234), then the digit length is 11

etc.

Yes but as stated these are "special numbers They are 3 to 4 digits long are such as 151, 1571, etc. They initial special services such as wake-up call, fault reporting, fault testing, last caller, voice mails etc. Most not used or needed by us but some we do. They HAVE to go out directly from the specific line. It used to work, does not now for some reason.

 

[Jonners59]

Hello,
Update firmware to 5.1.7
then reset to factory set

Hello Amygoda
Am ahead of you there.... I am on Software Version: 5.2.13(GW002) latest version

 

[Jonners59]

I've just upgraded both my 3102's - the latest firmware seems to be 5.2.13(GW002).

There is one trick that you can use to fully restore a previously saved SPA3102 configuration (rather than having to go through a printout and manually change all the parameters). Once you have a working config then save both the the router & voice configs to .htm files (as discussed above).

Edit both these files (in notepad) to replace the only occurrence of the line:
<form action="asipura.spa" method="POST">
with:
<form action="http://IP/admin/asipura.spa" method="POST">
[where IP is address of your SPA-3102]

And then resave as .htm again.

To restore the SPA3102 from the saved configuration, open the saved/edited file in a browser and hit "Submit All Changes" (you need to do this for each of the router & voice backup files).

Can save a lot of time if for example you want to quickly reconfigure a shelf-spare.

Chris

Chris
I had already done this, I am already on the latest version. Thanks for tip on how to store the config. It is rather an odd way of doing it, you solution will be priceless going forward....

I think the issue is that I hae a settings issue. Any tips welcome, please.

Echo I think is on the Gateway and I am wondering is the CLI is the 3CX given the line item on the log...

 

[leejor]

That sounds like a lot of guess work and trial and error. I'll see first if anyone can suggest some settings to try.

As far as echo, the defaults generally work well unless you have a phone line some distance from the central office, or, a lot of extra wiring/devices attached, that add to the capacitance of the line. Some devices, such as headsets will boost volume enough that you will hear an echo.

OK, that's interesting and noted. Seems a bit of a shortfall of the device to me. I would have expected the techies could have wporked out a simple no Ethernet/no registration go to call fail-over.... The nasty Chinese device I tried for all it's failings was able to do this. Now I know I nee to decide if it is better fto just plug the DECT phones straight in to the socket in the wall or the gateway, the res=ult seems to be the same, but at least the former i have a fail-over if Ethernet or registration fail, the down side is the DECT rings when I really only want to rely on the 3CX.

I'm not all that clear on what you are hoping that the 3102 will do. What is going to Failover to what? If Ethernet connectivity is lost, then the phone plugged into the FXS port won't be able to dial a VoIP provider number but will still be able to call out on the PSTN port using the 3102 dialplan. If power is lost the FXS port is connected directly to the PSTN line.

A call out over a 3CX trunk won't be able to reach the 3102 if Ethernet connectivity is lost, and as I said, in that case, it is up to 3CX to use a second choice route if registration is lost.

If you had a DECT set (not SIP), you could plug it into the FXS port of the 3102 and register that port as a 3CX extension. If the power failed the phone would be connected directly to the PSTN line, but, of course the base would be dead as the power would be out. You could just plug in a standard phone but , again, register the FXS as a 3Cx extension.

As far as CID, if the incoming 3CX logs show the calling parties number, then it should be showing up on the phones if they are capable of displaying it. Do you get the number if you make in internal, extension to extension call?

 

[Jonners59]

Leejor
I have nothing but the gateway on one line and one the other a fax too.... i am not too far from the excahnge and anyway we are on 21CN here with BT and VM is fibre to the door too so we are not really affected by distance. worked before. can you suggest what to put in the different parameters to test as a starting point. You mention "hot" but I do not know what that means.

Failover. .. you are looking at outbound. I am not iinterested in out bound for fail over as all calls are routed over my sip provider unless I wanted them to do otherwise and I have very easy auto and manual re routes and over ride solutions should I need them. No the issue is inbound only. From experience if a call comes in via the pstn as one would expect and for whatever reason the 3cx is down or not available then I want the calls to automatically re route to the fxo. I have had many years of experience where this has been needed. We have lost calls when we did not know we had an issue.

CLI: yes. Extn to extn works.... when I call from outside it says the name of the gateway and or the number of the acount... I.e. WORK and or 10001 ...... does that make sense?

 

[leejor]

have nothing but the gateway on one line and one the other a fax too.... i am not too far from the excahnge and anyway we are on 21CN here with BT and VM is fibre to the door too so we are not really affected by distance. worked before. can you suggest what to put in the different parameters to test as a starting point. You mention "hot" but I do not know what that means.

Hot means that the levels you are sending from your phones might be too high.
Have you tried any calls from more than one type of set?
Is it you that gets the echo, or the party you are calling?

Here is a site with some settings for the Linksys devices. Keep in mind that because of firmware differences, some terms may be different. Pay attention to the past bit about impedance.

http://www.aoakley.com/articles/2008-01-08.php

As far as failover. the 3102 contains a relay that is energized when there is power. When power is removed it simply connects the PSTN line (FXO) to the phone socket (FXS). That means that any phone plugged in there will work when the power fails.
I would seriously think about using the FXS port as a 3CX extension at least to do some testing with a different type of set.

With regards to CID, you haven't posted any logs of incoming calls showing CID, are you sure that 3CX is receiving it. Do you get CID from your VoIP provider? can you compare incoming calls from both sources?

I hope this link still works, it should allow you to download a PDF of the Admin manual for a number of Linksys devices. It has explanations of all of the fields as well as the default settings.

http://www.findthatpdf.com/search-94269605-hPDF/download-documents-SPA_Admn_3102.pdf.htm

 

[Jonners59]

have nothing but the gateway on one line and one the other a fax too.... i am not too far from the excahnge and anyway we are on 21CN here with BT and VM is fibre to the door too so we are not really affected by distance. worked before. can you suggest what to put in the different parameters to test as a starting point. You mention "hot" but I do not know what that means.

Hot means that the levels you are sending from your phones might be too high.
Have you tried any calls from more than one type of set?
Is it you that gets the echo, or the party you are calling?

I had an idea that is what you meant by hot....
Yes
As per first it is within the 3CX where the echo is heard. I.e. if I am calling an external number or I am answewring an external number then I hear the echo, not the person on the end of the line "outside" of the 3CX. I suspect a slight change might make the difference. What parameters would you suggest need changing and what should I put in from the current defailt setting(s)?

Here is a site with some settings for the Linksys devices. Keep in mind that because of firmware differences, some terms may be different. Pay attention to the past bit about impedance.

http://www.aoakley.com/articles/2008-01-08.php

Cheeky. This is the site I was following at the start and was told to ignore as it was for setting up an FXS, not for use on a 3CX....

As far as failover. the 3102 contains a relay that is energized when there is power. When power is removed it simply connects the PSTN line (FXO) to the phone socket (FXS). That means that any phone plugged in there will work when the power fails.
I would seriously think about using the FXS port as a 3CX extension at least to do some testing with a different type of set.

No I understand how it works now, just a bit dissapointed that the failover is not cleverer, especially given how complex and old it is.

With regards to CID, you haven't posted any logs of incoming calls showing CID, are you sure that 3CX is receiving it. Do you get CID from your VoIP provider? can you compare incoming calls from both sources?

No, I need to do that, send logs. I shall try later tonight when I get back if I can organise the family to make calls.....

I hope this link still works, it should allow you to download a PDF of the Admin manual for a number of Linksys devices. It has explanations of all of the fields as well as the default settings.

http://www.findthatpdf.com/search-94269605-hPDF/download-documents-SPA_Admn_3102.pdf.htm

I'll take a look, but these things scrae the pants off me!

The issue of my dial plan on the 3CX seems to have grown more complex. Just realised that ANY outbound call directed at going directly is now failing. I tried to make calls out from the FXS port on the gateway and all it did was call back in on the other gateway's inbound port account on the 3CX. B1oo6y confusing. I wonder if it is the Gateway dial plans. What does all this mean? It is the default settings. Could this be forcing the calls to fail.

(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)

I need to run more tests to work out what is happening. It is not as clear as some gateway information, so hard to follow. Oh, my little brain hurts.

 

[Jonners59]

OK, long awaited sys logs.
Bar below, all calls work fine, and I have zero issues with them, even echo seems to have stopped, I have changed the FXO impedance setting on "Line"

Issues needing resolving are CLI and the below forcing calls from the 3CX to the PSTN via the Gateway FXO/PSTN, items one and two, and by-passing the 3CX and making a direct call via the gateway FXS to FXO/PSTN, which seem to route back in, all of which imply to me that the Gateway does not want to send any calls to the FSO/PSTN, so routes them back or via the default alternative route.... dial plan on gateway issue?????

Test 1:
Outbound call - forcing the call to go out via the FXO with the use of "4" in the dial plan: all calls starting with "4" will route to FXO Office. Note the call does not go out as instructed but relies on the fall-back route, which is via the SIP provider.....

28-Nov-2013 13:53:26.177   [CM503008]: Call(C:108): Call is terminated
28-Nov-2013 13:53:26.177   L:108.3[Line:10005>>07770740133]: Terminating targets, reason: SIP ;cause=200 ;text="Call terminated on user request"
28-Nov-2013 13:53:26.177   Blocking refers for the Call(C:108)
28-Nov-2013 13:53:26.177   Call(C:108) is terminated
28-Nov-2013 13:53:26.177   ~Target=PSTNline:07770740133@(Ln.10002@HOME)
28-Nov-2013 13:53:26.177   ~Route=Dev:sip:[email protected]:5062
28-Nov-2013 13:53:26.177   ~Target=VoIPline:07770740133@(Ln.10005@VoIPTalk)
28-Nov-2013 13:53:26.177   ~Route=Dev:sip:[email protected]:5060
28-Nov-2013 13:53:26.177   L:108.3[Line:10005>>07770740133]: Terminating targets, reason: SIP ;cause=487 ;text="Request Terminated"
28-Nov-2013 13:53:26.142   Reason: SIP ;cause=487 ;text="Request Terminated"
28-Nov-2013 13:53:26.142   L:108.1[Extn]: Terminating targets, reason: SIP ;cause=487 ;text="Request Terminated"
28-Nov-2013 13:53:26.142   Leg L:108.1[Extn] is terminated: Cause: CANCEL from 192.168.1.90:5060
28-Nov-2013 13:53:26.124   L:108.1[Extn] got Terminated Recv Req CANCEL from 192.168.1.90:5060 tid=1a9b1253550914bb841cdfb28ae37d9a Call-ID=4154191590@192_168_1_90:
CANCEL sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK1a9b1253550914bb841cdfb28ae37d9a;rport=5060
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected];user=phone>
From: "Baroni Limited"<sip:[email protected]>;tag=606715209
Call-ID: 4154191590@192_168_1_90
CSeq: 3 CANCEL
Proxy-Authorization: Digest username="200", realm="3CXPhoneSystem", algorithm=MD5, uri="sip:[email protected];user=phone", nonce="414d535c08a7dbc819:a042ad3c80245cd1887c9c8097da49e0", response="a75d7a02ef2c87807fc3945908080138"
User-Agent: N300 IP/42.075.00.000.000
Content-Length: 0
28-Nov-2013 13:53:26.124   Terminated from "Baroni Limited"<sip:[email protected]>;tag=606715209 to <sip:[email protected];user=phone>;tag=ca38f55b; reason: RemoteBye
28-Nov-2013 13:53:26.124   L:108.1[Extn] Sending: OnSendResp Send 487/INVITE from 0.0.0.0:0 tid=1a9b1253550914bb841cdfb28ae37d9a Call-ID=4154191590@192_168_1_90:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK1a9b1253550914bb841cdfb28ae37d9a;rport=5060
To: <sip:[email protected];user=phone>;tag=ca38f55b
From: "Baroni Limited"<sip:[email protected]>;tag=606715209
Call-ID: 4154191590@192_168_1_90
CSeq: 3 INVITE
Content-Length: 0
28-Nov-2013 13:53:26.124   SendMsg from <sip:[email protected];user=phone>;tag=ca38f55b to "Baroni Limited"<sip:[email protected]>;tag=606715209
28-Nov-2013 13:53:26.124   L:108.1[Extn] Sending: OnSendResp Send 200/CANCEL from 0.0.0.0:0 tid=1a9b1253550914bb841cdfb28ae37d9a Call-ID=4154191590@192_168_1_90:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK1a9b1253550914bb841cdfb28ae37d9a;rport=5060
Contact: <sip:[email protected];user=phone>
To: <sip:[email protected];user=phone>;tag=ca38f55b
From: "Baroni Limited"<sip:[email protected]>;tag=606715209
Call-ID: 4154191590@192_168_1_90
CSeq: 3 CANCEL
Content-Length: 0
28-Nov-2013 13:53:26.124   SendMsg from <sip:[email protected];user=phone>;tag=ca38f55b to "Baroni Limited"<sip:[email protected]>;tag=606715209
28-Nov-2013 13:53:23.371   L:108.3[Line:10005>>07770740133] got EarlyMedia Recv 183/INVITE from 77.240.48.201:5065 tid=c81dc97bdb039053 Call-ID=ZmExMzFhNTQyZmVjZmExNTc0MGE4OGRkNjFhMjNmNzA.:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.51:5060;received=82.14.176.51;branch=z9hG4bK-d8754z-c81dc97bdb039053-1---d8754z-;rport=5060
Record-Route: <sip:77.240.48.94;lr=on;ftag=b43a4b1c>
Record-Route: <sip:[email protected];r2=on;lr=on;ftag=b43a4b1c>
Record-Route: <sip:[email protected]:5065;r2=on;lr=on;ftag=b43a4b1c>
Contact: <sip:[email protected]>
To: <sip:[email protected]:5060>;tag=as12baa533
From: "Baroni Limited"<sip:[email protected]:5060>;tag=b43a4b1c
Call-ID: ZmExMzFhNTQyZmVjZmExNTc0MGE4OGRkNjFhMjNmNzA.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Content-Type: application/sdp
Server: voip
Supported: replaces
Content-Length: 309
P-RTP-Proxy: Yes

v=0
o=voip 59551206 59551206 IN IP4 77.240.54.23
s=voip
c=IN IP4 77.240.48.209
t=0 0
m=audio 39508 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=nortpproxy:yes
28-Nov-2013 13:53:23.371   EarlyMedia from <sip:[email protected]:5060>;tag=as12baa533 to "Baroni Limited"<sip:[email protected]:5060>;tag=b43a4b1c
28-Nov-2013 13:53:23.371   Provisional response arrived for session 80547 of Leg L:108.3[Line:10005>>07770740133]
28-Nov-2013 13:53:23.371   L:108.3[Line:10005>>07770740133] got Provisional Recv 183/INVITE from 77.240.48.201:5065 tid=c81dc97bdb039053 Call-ID=ZmExMzFhNTQyZmVjZmExNTc0MGE4OGRkNjFhMjNmNzA.:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.51:5060;received=82.14.176.51;branch=z9hG4bK-d8754z-c81dc97bdb039053-1---d8754z-;rport=5060
Record-Route: <sip:77.240.48.94;lr=on;ftag=b43a4b1c>
Record-Route: <sip:[email protected];r2=on;lr=on;ftag=b43a4b1c>
Record-Route: <sip:[email protected]:5065;r2=on;lr=on;ftag=b43a4b1c>
Contact: <sip:[email protected]>
To: <sip:[email protected]:5060>;tag=as12baa533
From: "Baroni Limited"<sip:[email protected]:5060>;tag=b43a4b1c
Call-ID: ZmExMzFhNTQyZmVjZmExNTc0MGE4OGRkNjFhMjNmNzA.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Content-Type: application/sdp
Server: voip
Supported: replaces
Content-Length: 309
P-RTP-Proxy: Yes

v=0
o=voip 59551206 59551206 IN IP4 77.240.54.23
s=voip
c=IN IP4 77.240.48.209
t=0 0
m=audio 39508 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=nortpproxy:yes
28-Nov-2013 13:53:23.371   Provisional(183) from <sip:[email protected]:5060>;tag=as12baa533 to "Baroni Limited"<sip:[email protected]:5060>;tag=b43a4b1c
28-Nov-2013 13:53:23.371   L:108.1[Extn] Sending: OnSendResp Send 180/INVITE from 0.0.0.0:0 tid=1a9b1253550914bb841cdfb28ae37d9a Call-ID=4154191590@192_168_1_90:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK1a9b1253550914bb841cdfb28ae37d9a;rport=5060
Contact: <sip:[email protected];user=phone>
To: <sip:[email protected];user=phone>;tag=ca38f55b
From: "Baroni Limited"<sip:[email protected]>;tag=606715209
Call-ID: 4154191590@192_168_1_90
CSeq: 3 INVITE
Content-Length: 0
28-Nov-2013 13:53:23.371   SendMsg from <sip:[email protected];user=phone>;tag=ca38f55b to "Baroni Limited"<sip:[email protected]>;tag=606715209
28-Nov-2013 13:53:23.371   Provisional response arrived for session 80547 of Leg L:108.3[Line:10005>>07770740133]
28-Nov-2013 13:53:23.371   L:108.3[Line:10005>>07770740133] got Provisional Recv 180/INVITE from 77.240.48.201:5065 tid=c81dc97bdb039053 Call-ID=ZmExMzFhNTQyZmVjZmExNTc0MGE4OGRkNjFhMjNmNzA.:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.51:5060;received=82.14.176.51;branch=z9hG4bK-d8754z-c81dc97bdb039053-1---d8754z-;rport=5060
Record-Route: <sip:77.240.48.94;lr=on;ftag=b43a4b1c>
Record-Route: <sip:[email protected];r2=on;lr=on;ftag=b43a4b1c>
Record-Route: <sip:[email protected]:5065;r2=on;lr=on;ftag=b43a4b1c>
Contact: <sip:[email protected]>
To: <sip:[email protected]:5060>;tag=as12baa533
From: "Baroni Limited"<sip:[email protected]:5060>;tag=b43a4b1c
Call-ID: ZmExMzFhNTQyZmVjZmExNTc0MGE4OGRkNjFhMjNmNzA.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Server: voip
Supported: replaces
Content-Length: 0
28-Nov-2013 13:53:23.371   Provisional(180) from <sip:[email protected]:5060>;tag=as12baa533 to "Baroni Limited"<sip:[email protected]:5060>;tag=b43a4b1c
28-Nov-2013 13:53:23.371   [CM505003]: Provider:[VoIPTalk] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] PBX contact: [sip:[email protected]:5060]
28-Nov-2013 13:53:23.371   [CM503002]: Call(C:108): Alerting Line:10005>>07770740133 by contact <sip:[email protected]:5060>
28-Nov-2013 13:53:23.337   Phonebook entry not found for number 07770740133
28-Nov-2013 13:53:23.337   Looking for phone number 07770740133 in tenant's 'default' phonebook
28-Nov-2013 13:53:23.337   UacSession 80547 has formed leg L:108.3[Line:10005>>07770740133]
28-Nov-2013 13:53:23.337   Answer from <sip:[email protected]:5060>;tag=as12baa533 to "Baroni Limited"<sip:[email protected]:5060>;tag=b43a4b1c
28-Nov-2013 13:53:13.220   L:108.2[Line:10001>>07770740133]: Terminating targets, reason: SIP ;cause=404 ;text="Not Found"
28-Nov-2013 13:53:13.220   Leg L:108.2[Line:10001>>07770740133] is terminated: Cause: 404 Not Found/INVITE from 192.168.1.95:5062
28-Nov-2013 13:53:13.220   L:108.2[Line:10001>>07770740133] got Terminated Recv 404/INVITE from 192.168.1.95:5062 tid=242e472760152e25 Call-ID=ZjBhMzM1MTE1OTU1M2MxY2NkODhlMTc1YjFhYTFhMmY.:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.51:5060;branch=z9hG4bK-d8754z-242e472760152e25-1---d8754z-
To: <sip:[email protected]:5062>;tag=8a2791dbc65d347i0
From: "Baroni Limited"<sip:[email protected]:5060>;tag=8437294a
Call-ID: ZjBhMzM1MTE1OTU1M2MxY2NkODhlMTc1YjFhYTFhMmY.
CSeq: 1 INVITE
Server: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 0
28-Nov-2013 13:53:13.220   Terminated from <sip:[email protected]:5062>;tag=8a2791dbc65d347i0 to "Baroni Limited"<sip:[email protected]:5060>;tag=8437294a; reason: Error
28-Nov-2013 13:53:13.220   ~Target=PSTNline:07770740133@(Ln.10001@OFFICE)
28-Nov-2013 13:53:13.220   [CM503025]: Call(C:108): Calling T:Line:10005>>07770740133@[Dev:sip:[email protected]:5060] for L:108.1[Extn]
28-Nov-2013 13:53:13.220   Route to L:108.3[Line:10005>>07770740133] sends Invite-OUT Send Req INVITE from 0.0.0.0:0 tid=16428373312bc565 Call-ID=ZmExMzFhNTQyZmVjZmExNTc0MGE4OGRkNjFhMjNmNzA.:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/ ;branch=z9hG4bK-d8754z-16428373312bc565-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]:5060>
From: "Baroni Limited"<sip:[email protected]:5060>;tag=b43a4b1c
Call-ID: ZmExMzFhNTQyZmVjZmExNTc0MGE4OGRkNjFhMjNmNzA.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
Content-Length: 281

v=0
o=3cxPS 164131504128 281286803457 IN IP4 82.14.176.51
s=3cxPS Audio call
c=IN IP4 82.14.176.51
t=0 0
m=audio 9002 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
28-Nov-2013 13:53:13.216   Dev(1):[sip:[email protected]:5060 / 844231085]: PBX contact is resolved to public IP: <sip:[email protected]:5060>
28-Nov-2013 13:53:13.106   Outbound URI is used: sip:77.240.48.201:5065
28-Nov-2013 13:53:13.106   Added leg L:108.3[Line:10005>>07770740133]
28-Nov-2013 13:53:13.106   ~Route=Dev:sip:[email protected]:5062
28-Nov-2013 13:53:13.106   Call to T:Line:10001>>07770740133@[Dev:sip:[email protected]:5062] from L:108.1[Extn] failed, cause: Cause: 404 Not Found/INVITE from 192.168.1.95:5062
28-Nov-2013 13:53:13.101   L:108.2[Line:10001>>07770740133] got Failure: Failure Recv 404/INVITE from 192.168.1.95:5062 tid=242e472760152e25 Call-ID=ZjBhMzM1MTE1OTU1M2MxY2NkODhlMTc1YjFhYTFhMmY.:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.51:5060;branch=z9hG4bK-d8754z-242e472760152e25-1---d8754z-
To: <sip:[email protected]:5062>;tag=8a2791dbc65d347i0
From: "Baroni Limited"<sip:[email protected]:5060>;tag=8437294a
Call-ID: ZjBhMzM1MTE1OTU1M2MxY2NkODhlMTc1YjFhYTFhMmY.
CSeq: 1 INVITE
Server: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 0
28-Nov-2013 13:53:13.101   [CM503003]: Call(C:108): Call to <sip:[email protected]:5062> has failed; Cause: 404 Not Found/INVITE from 192.168.1.95:5062
28-Nov-2013 13:53:13.101   Session 80542 has failed in leg L:108.2[Line:10001>>07770740133] ; Cause: 404 Not Found/INVITE from 192.168.1.95:5062
28-Nov-2013 13:53:13.101   Failure from <sip:[email protected]:5062>;tag=8a2791dbc65d347i0 to "Baroni Limited"<sip:[email protected]:5060>;tag=8437294a
28-Nov-2013 13:53:13.018   [CM503025]: Call(C:108): Calling T:Line:10001>>07770740133@[Dev:sip:[email protected]:5062] for L:108.1[Extn]
28-Nov-2013 13:53:13.018   Route to L:108.2[Line:10001>>07770740133] sends Invite-OUT Send Req INVITE from 0.0.0.0:0 tid=422fe65f4f6fa418 Call-ID=ZjBhMzM1MTE1OTU1M2MxY2NkODhlMTc1YjFhYTFhMmY.:
INVITE sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/ ;branch=z9hG4bK-d8754z-422fe65f4f6fa418-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]:5062>
From: "Baroni Limited"<sip:[email protected]:5060>;tag=8437294a
Call-ID: ZjBhMzM1MTE1OTU1M2MxY2NkODhlMTc1YjFhYTFhMmY.
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 158091706368 402820956161 IN IP4 192.168.1.51
s=3cxPS Audio call
c=IN IP4 192.168.1.51
t=0 0
m=audio 7016 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
28-Nov-2013 13:53:12.996   Outbound URI is used: sip:[email protected]:5062
28-Nov-2013 13:53:12.981   Added leg L:108.2[Line:10001>>07770740133]
28-Nov-2013 13:53:12.950   [Flow] Call(C:108): making call from L:108.1[Extn] to T:Line:10002>>07770740133@[Dev:sip:[email protected]:5062]
28-Nov-2013 13:53:12.950   [CM503027]: Call(C:108): From: Extn:200 ("Baroni Limited" <sip:[email protected]:5060>)  to  T:Line:10002>>07770740133@[Dev:sip:[email protected]:5062]
28-Nov-2013 13:53:12.950   [CM503004]: Call(C:108): Route 3: from L:108.1[Extn] to T:Line:10002>>07770740133@[Dev:sip:[email protected]:5062]
28-Nov-2013 13:53:12.950   Line limit check: Current # of calls for line Lc:10002(@HOME[<sip:[email protected]:5062>]) is 0; limit is 1
28-Nov-2013 13:53:12.950   [Flow] Call(C:108): making call from L:108.1[Extn] to T:Line:10005>>07770740133@[Dev:sip:[email protected]:5060]
28-Nov-2013 13:53:12.950   [CM503027]: Call(C:108): From: Extn:200 ("Baroni Limited" <sip:[email protected]:5060>)  to  T:Line:10005>>07770740133@[Dev:sip:[email protected]:5060]
28-Nov-2013 13:53:12.950   [CM503004]: Call(C:108): Route 2: from L:108.1[Extn] to T:Line:10005>>07770740133@[Dev:sip:[email protected]:5060]
28-Nov-2013 13:53:12.950   Line limit check: Current # of calls for line Lc:10005(@VoIPTalk[<sip:[email protected]:5060>]) is 0; limit is 2
28-Nov-2013 13:53:12.949   [Flow] Call(C:108): making call from L:108.1[Extn] to T:Line:10001>>07770740133@[Dev:sip:[email protected]:5062]
28-Nov-2013 13:53:12.949   [CM503027]: Call(C:108): From: Extn:200 ("Baroni Limited" <sip:[email protected]:5060>)  to  T:Line:10001>>07770740133@[Dev:sip:[email protected]:5062]
28-Nov-2013 13:53:12.949   [CM503004]: Call(C:108): Route 1: from L:108.1[Extn] to T:Line:10001>>07770740133@[Dev:sip:[email protected]:5062]
28-Nov-2013 13:53:12.949   Line limit check: Current # of calls for line Lc:10001(@OFFICE[<sip:[email protected]:5062>]) is 0; limit is 1
28-Nov-2013 13:53:12.949   Call(C:108): Call from Extn:200 to 407770740133 matches outbound rule 'PSTN Baroni'
28-Nov-2013 13:53:12.949   [Flow] Call(C:108): has built target endpoint: Out#:>>Rule{PSTN Baroni}>>407770740133 for call from L:108.1[Extn]
28-Nov-2013 13:53:12.949   [Flow] Target endpoint for 407770740133 is Out#:>>Rule{PSTN Baroni}>>407770740133
28-Nov-2013 13:53:12.949   Selected prefix: 4
28-Nov-2013 13:53:12.949   Looking for outbound rule: dialed = [407770740133], processed: [407770740133]; from-ext:
28-Nov-2013 13:53:12.949   [Flow] Building target endpoint to 407770740133 from "Baroni Limited" <sip:[email protected]:5060>
28-Nov-2013 13:53:12.949   [CM503010]: Call(C:108): Making route(s) from Extn:200 to <sip:[email protected]:5060>
28-Nov-2013 13:53:12.949   Remote SDP is set for leg L:108.1[Extn]
28-Nov-2013 13:53:12.949   OnOffer from "Baroni Limited"<sip:[email protected]>;tag=606715209
28-Nov-2013 13:53:12.947   [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]
28-Nov-2013 13:53:12.923   Inbound DID: ''; Phonebook Name: ''
28-Nov-2013 13:53:12.923   [CM500002]: Call(C:108): Info on incoming INVITE from Extn:200:
Invite-IN Recv Req INVITE from 192.168.1.90:5060 tid=1a9b1253550914bb841cdfb28ae37d9a Call-ID=4154191590@192_168_1_90:
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK1a9b1253550914bb841cdfb28ae37d9a;rport=5060
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected];user=phone>
From: "Baroni Limited"<sip:[email protected]>;tag=606715209
Call-ID: 4154191590@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="414d535c08a7dbc819:a042ad3c80245cd1887c9c8097da49e0",response="c0e745b882a2aa209a076aaf9584fa7c"
Supported: replaces
User-Agent: N300 IP/42.075.00.000.000
Allow-Events: message-summary, refer, ua-profile, talk
Content-Length: 379

v=0
o=200 5008 4 IN IP4 192.168.1.90
s=Mapping
c=IN IP4 192.168.1.90
t=0 0
m=audio 5008 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
28-Nov-2013 13:53:12.923   [CM503001]: Call(C:108): Incoming call from Extn:200 to <sip:[email protected]:5060>

Test 2:
The same as above but a different plan. Forcing a call out over the Gateway FXO/PSTN for local services - all numbers starting with a "1" (ONE).

28-Nov-2013 14:12:12.464   L:110.4[Line:10005>>07770740133] got EarlyMedia Recv 183/INVITE from 77.240.48.201:5065 tid=954847406b6c670f Call-ID=YTdjYWFmZTQ0OWU5Yzk0NDE0Mjc2YzY3YTUwMWIwZWQ.:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.51:5060;received=82.14.176.51;branch=z9hG4bK-d8754z-954847406b6c670f-1---d8754z-;rport=5060
Record-Route: <sip:77.240.48.94;lr=on;ftag=84234211>
Record-Route: <sip:[email protected];r2=on;lr=on;ftag=84234211>
Record-Route: <sip:[email protected]:5065;r2=on;lr=on;ftag=84234211>
Contact: <sip:[email protected]:5061>
To: <sip:[email protected]:5060>;tag=as2049ca55
From: "Baroni Limited"<sip:[email protected]:5060>;tag=84234211
Call-ID: YTdjYWFmZTQ0OWU5Yzk0NDE0Mjc2YzY3YTUwMWIwZWQ.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Content-Type: application/sdp
Server: voip
Supported: replaces
Content-Length: 311
P-RTP-Proxy: Yes

v=0
o=voip 148428099 148428099 IN IP4 77.240.54.13
s=voip
c=IN IP4 77.240.48.209
t=0 0
m=audio 47992 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=nortpproxy:yes
28-Nov-2013 14:12:12.464   EarlyMedia from <sip:[email protected]:5060>;tag=as2049ca55 to "Baroni Limited"<sip:[email protected]:5060>;tag=84234211
28-Nov-2013 14:12:12.464   Provisional response arrived for session 80854 of Leg L:110.4[Line:10005>>07770740133]
28-Nov-2013 14:12:12.464   L:110.4[Line:10005>>07770740133] got Provisional Recv 183/INVITE from 77.240.48.201:5065 tid=954847406b6c670f Call-ID=YTdjYWFmZTQ0OWU5Yzk0NDE0Mjc2YzY3YTUwMWIwZWQ.:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.51:5060;received=82.14.176.51;branch=z9hG4bK-d8754z-954847406b6c670f-1---d8754z-;rport=5060
Record-Route: <sip:77.240.48.94;lr=on;ftag=84234211>
Record-Route: <sip:[email protected];r2=on;lr=on;ftag=84234211>
Record-Route: <sip:[email protected]:5065;r2=on;lr=on;ftag=84234211>
Contact: <sip:[email protected]:5061>
To: <sip:[email protected]:5060>;tag=as2049ca55
From: "Baroni Limited"<sip:[email protected]:5060>;tag=84234211
Call-ID: YTdjYWFmZTQ0OWU5Yzk0NDE0Mjc2YzY3YTUwMWIwZWQ.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Content-Type: application/sdp
Server: voip
Supported: replaces
Content-Length: 311
P-RTP-Proxy: Yes

v=0
o=voip 148428099 148428099 IN IP4 77.240.54.13
s=voip
c=IN IP4 77.240.48.209
t=0 0
m=audio 47992 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=nortpproxy:yes
28-Nov-2013 14:12:12.464   Provisional(183) from <sip:[email protected]:5060>;tag=as2049ca55 to "Baroni Limited"<sip:[email protected]:5060>;tag=84234211
28-Nov-2013 14:12:12.457   L:110.1[Extn] Sending: OnSendResp Send 180/INVITE from 0.0.0.0:0 tid=2245037638dcc2ba22f67b44359c564d Call-ID=1786777608@192_168_1_90:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK2245037638dcc2ba22f67b44359c564d;rport=5060
Contact: <sip:[email protected];user=phone>
To: <sip:[email protected];user=phone>;tag=7a0ca106
From: "Baroni Limited"<sip:[email protected]>;tag=1328908937
Call-ID: 1786777608@192_168_1_90
CSeq: 3 INVITE
Content-Length: 0
28-Nov-2013 14:12:12.457   SendMsg from <sip:[email protected];user=phone>;tag=7a0ca106 to "Baroni Limited"<sip:[email protected]>;tag=1328908937
28-Nov-2013 14:12:12.457   Provisional response arrived for session 80854 of Leg L:110.4[Line:10005>>07770740133]
28-Nov-2013 14:12:12.457   L:110.4[Line:10005>>07770740133] got Provisional Recv 180/INVITE from 77.240.48.201:5065 tid=954847406b6c670f Call-ID=YTdjYWFmZTQ0OWU5Yzk0NDE0Mjc2YzY3YTUwMWIwZWQ.:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.51:5060;received=82.14.176.51;branch=z9hG4bK-d8754z-954847406b6c670f-1---d8754z-;rport=5060
Record-Route: <sip:77.240.48.94;lr=on;ftag=84234211>
Record-Route: <sip:[email protected];r2=on;lr=on;ftag=84234211>
Record-Route: <sip:[email protected]:5065;r2=on;lr=on;ftag=84234211>
Contact: <sip:[email protected]:5061>
To: <sip:[email protected]:5060>;tag=as2049ca55
From: "Baroni Limited"<sip:[email protected]:5060>;tag=84234211
Call-ID: YTdjYWFmZTQ0OWU5Yzk0NDE0Mjc2YzY3YTUwMWIwZWQ.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Server: voip
Supported: replaces
Content-Length: 0
28-Nov-2013 14:12:12.457   Provisional(180) from <sip:[email protected]:5060>;tag=as2049ca55 to "Baroni Limited"<sip:[email protected]:5060>;tag=84234211
28-Nov-2013 14:12:12.457   [CM505003]: Provider:[VoIPTalk] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] PBX contact: [sip:[email protected]:5060]
28-Nov-2013 14:12:12.457   [CM503002]: Call(C:110): Alerting Line:10005>>07770740133 by contact <sip:[email protected]:5060>
28-Nov-2013 14:12:12.444   Phonebook entry not found for number 07770740133
28-Nov-2013 14:12:12.444   Looking for phone number 07770740133 in tenant's 'default' phonebook
28-Nov-2013 14:12:12.444   UacSession 80854 has formed leg L:110.4[Line:10005>>07770740133]
28-Nov-2013 14:12:12.444   Answer from <sip:[email protected]:5060>;tag=as2049ca55 to "Baroni Limited"<sip:[email protected]:5060>;tag=84234211
28-Nov-2013 14:12:02.080   L:110.3[Line:10001>>07770740133]: Terminating targets, reason: SIP ;cause=404 ;text="Not Found"
28-Nov-2013 14:12:02.080   Leg L:110.3[Line:10001>>07770740133] is terminated: Cause: 404 Not Found/INVITE from 192.168.1.95:5062
28-Nov-2013 14:12:02.080   L:110.3[Line:10001>>07770740133] got Terminated Recv 404/INVITE from 192.168.1.95:5062 tid=d22b0b22fa503a35 Call-ID=NWI0Zjk3NmZjNzEwNWI4NzlmNjk3Zjg3NTUyZTgyNzQ.:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.51:5060;branch=z9hG4bK-d8754z-d22b0b22fa503a35-1---d8754z-
To: <sip:[email protected]:5062>;tag=8a2791dbc65d347i0
From: "Baroni Limited"<sip:[email protected]:5060>;tag=7d477126
Call-ID: NWI0Zjk3NmZjNzEwNWI4NzlmNjk3Zjg3NTUyZTgyNzQ.
CSeq: 1 INVITE
Server: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 0
28-Nov-2013 14:12:02.080   Terminated from <sip:[email protected]:5062>;tag=8a2791dbc65d347i0 to "Baroni Limited"<sip:[email protected]:5060>;tag=7d477126; reason: Error
28-Nov-2013 14:12:02.080   ~Target=PSTNline:07770740133@(Ln.10001@OFFICE)
28-Nov-2013 14:12:02.080   [CM503025]: Call(C:110): Calling T:Line:10005>>07770740133@[Dev:sip:[email protected]:5060] for L:110.1[Extn]
28-Nov-2013 14:12:02.080   Route to L:110.4[Line:10005>>07770740133] sends Invite-OUT Send Req INVITE from 0.0.0.0:0 tid=bd1c9235e116582e Call-ID=YTdjYWFmZTQ0OWU5Yzk0NDE0Mjc2YzY3YTUwMWIwZWQ.:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/ ;branch=z9hG4bK-d8754z-bd1c9235e116582e-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]:5060>
From: "Baroni Limited"<sip:[email protected]:5060>;tag=84234211
Call-ID: YTdjYWFmZTQ0OWU5Yzk0NDE0Mjc2YzY3YTUwMWIwZWQ.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
Content-Length: 281

v=0
o=3cxPS 422970392576 369820172289 IN IP4 82.14.176.51
s=3cxPS Audio call
c=IN IP4 82.14.176.51
t=0 0
m=audio 9004 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
28-Nov-2013 14:12:02.080   Dev(1):[sip:[email protected]:5060 / 844231085]: PBX contact is resolved to public IP: <sip:[email protected]:5060>
28-Nov-2013 14:12:01.978   Outbound URI is used: sip:77.240.48.201:5065
28-Nov-2013 14:12:01.978   Added leg L:110.4[Line:10005>>07770740133]
28-Nov-2013 14:12:01.978   ~Route=Dev:sip:[email protected]:5062
28-Nov-2013 14:12:01.978   Call to T:Line:10001>>07770740133@[Dev:sip:[email protected]:5062] from L:110.1[Extn] failed, cause: Cause: 404 Not Found/INVITE from 192.168.1.95:5062
28-Nov-2013 14:12:01.973   L:110.3[Line:10001>>07770740133] got Failure: Failure Recv 404/INVITE from 192.168.1.95:5062 tid=d22b0b22fa503a35 Call-ID=NWI0Zjk3NmZjNzEwNWI4NzlmNjk3Zjg3NTUyZTgyNzQ.:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.51:5060;branch=z9hG4bK-d8754z-d22b0b22fa503a35-1---d8754z-
To: <sip:[email protected]:5062>;tag=8a2791dbc65d347i0
From: "Baroni Limited"<sip:[email protected]:5060>;tag=7d477126
Call-ID: NWI0Zjk3NmZjNzEwNWI4NzlmNjk3Zjg3NTUyZTgyNzQ.
CSeq: 1 INVITE
Server: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 0
28-Nov-2013 14:12:01.973   [CM503003]: Call(C:110): Call to <sip:[email protected]:5062> has failed; Cause: 404 Not Found/INVITE from 192.168.1.95:5062
28-Nov-2013 14:12:01.973   Session 80843 has failed in leg L:110.3[Line:10001>>07770740133] ; Cause: 404 Not Found/INVITE from 192.168.1.95:5062
28-Nov-2013 14:12:01.973   Failure from <sip:[email protected]:5062>;tag=8a2791dbc65d347i0 to "Baroni Limited"<sip:[email protected]:5060>;tag=7d477126
28-Nov-2013 14:12:01.876   L:110.2[Line:10002>>189907770740133]: Terminating targets, reason: SIP ;cause=404 ;text="Not Found"
28-Nov-2013 14:12:01.876   Leg L:110.2[Line:10002>>189907770740133] is terminated: Cause: 404 Not Found/INVITE from 192.168.1.96:5062
28-Nov-2013 14:12:01.876   L:110.2[Line:10002>>189907770740133] got Terminated Recv 404/INVITE from 192.168.1.96:5062 tid=57136c67112f1424 Call-ID=M2VhZGUzODBiMzFmYTQ3ODM1NDI2ZTZjODRmZTRhYjU.:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.51:5060;branch=z9hG4bK-d8754z-57136c67112f1424-1---d8754z-
To: <sip:[email protected]:5062>;tag=7f4e78be1523e282i0
From: "Baroni Limited"<sip:[email protected]:5060>;tag=d0066266
Call-ID: M2VhZGUzODBiMzFmYTQ3ODM1NDI2ZTZjODRmZTRhYjU.
CSeq: 1 INVITE
Server: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 0
28-Nov-2013 14:12:01.876   Terminated from <sip:[email protected]:5062>;tag=7f4e78be1523e282i0 to "Baroni Limited"<sip:[email protected]:5060>;tag=d0066266; reason: Error
28-Nov-2013 14:12:01.876   ~Target=PSTNline:189907770740133@(Ln.10002@HOME)
28-Nov-2013 14:12:01.876   [CM503025]: Call(C:110): Calling T:Line:10001>>07770740133@[Dev:sip:[email protected]:5062] for L:110.1[Extn]
28-Nov-2013 14:12:01.876   Route to L:110.3[Line:10001>>07770740133] sends Invite-OUT Send Req INVITE from 0.0.0.0:0 tid=57783333d63b6e57 Call-ID=NWI0Zjk3NmZjNzEwNWI4NzlmNjk3Zjg3NTUyZTgyNzQ.:
INVITE sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/ ;branch=z9hG4bK-d8754z-57783333d63b6e57-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]:5062>
From: "Baroni Limited"<sip:[email protected]:5060>;tag=7d477126
Call-ID: NWI0Zjk3NmZjNzEwNWI4NzlmNjk3Zjg3NTUyZTgyNzQ.
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 416393723904 302476427265 IN IP4 192.168.1.51
s=3cxPS Audio call
c=IN IP4 192.168.1.51
t=0 0
m=audio 7028 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
28-Nov-2013 14:12:01.856   Outbound URI is used: sip:[email protected]:5062
28-Nov-2013 14:12:01.856   Added leg L:110.3[Line:10001>>07770740133]
28-Nov-2013 14:12:01.856   ~Route=Dev:sip:[email protected]:5062
28-Nov-2013 14:12:01.856   Call to T:Line:10002>>189907770740133@[Dev:sip:[email protected]:5062] from L:110.1[Extn] failed, cause: Cause: 404 Not Found/INVITE from 192.168.1.96:5062
28-Nov-2013 14:12:01.849   L:110.2[Line:10002>>189907770740133] got Failure: Failure Recv 404/INVITE from 192.168.1.96:5062 tid=57136c67112f1424 Call-ID=M2VhZGUzODBiMzFmYTQ3ODM1NDI2ZTZjODRmZTRhYjU.:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.51:5060;branch=z9hG4bK-d8754z-57136c67112f1424-1---d8754z-
To: <sip:[email protected]:5062>;tag=7f4e78be1523e282i0
From: "Baroni Limited"<sip:[email protected]:5060>;tag=d0066266
Call-ID: M2VhZGUzODBiMzFmYTQ3ODM1NDI2ZTZjODRmZTRhYjU.
CSeq: 1 INVITE
Server: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 0
28-Nov-2013 14:12:01.849   [CM503003]: Call(C:110): Call to <sip:[email protected]:5062> has failed; Cause: 404 Not Found/INVITE from 192.168.1.96:5062
28-Nov-2013 14:12:01.849   Session 80840 has failed in leg L:110.2[Line:10002>>189907770740133] ; Cause: 404 Not Found/INVITE from 192.168.1.96:5062
28-Nov-2013 14:12:01.849   Failure from <sip:[email protected]:5062>;tag=7f4e78be1523e282i0 to "Baroni Limited"<sip:[email protected]:5060>;tag=d0066266
28-Nov-2013 14:12:01.734   [CM503025]: Call(C:110): Calling T:Line:10002>>189907770740133@[Dev:sip:[email protected]:5062] for L:110.1[Extn]
28-Nov-2013 14:12:01.734   Route to L:110.2[Line:10002>>189907770740133] sends Invite-OUT Send Req INVITE from 0.0.0.0:0 tid=5e06e74df5165961 Call-ID=M2VhZGUzODBiMzFmYTQ3ODM1NDI2ZTZjODRmZTRhYjU.:
INVITE sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/ ;branch=z9hG4bK-d8754z-5e06e74df5165961-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]:5062>
From: "Baroni Limited"<sip:[email protected]:5060>;tag=d0066266
Call-ID: M2VhZGUzODBiMzFmYTQ3ODM1NDI2ZTZjODRmZTRhYjU.
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 407568908288 175489679361 IN IP4 192.168.1.51
s=3cxPS Audio call
c=IN IP4 192.168.1.51
t=0 0
m=audio 7026 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
28-Nov-2013 14:12:01.705   Outbound URI is used: sip:[email protected]:5062
28-Nov-2013 14:12:01.693   Added leg L:110.2[Line:10002>>189907770740133]
28-Nov-2013 14:12:01.660   [Flow] Call(C:110): making call from L:110.1[Extn] to T:Line:10005>>07770740133@[Dev:sip:[email protected]:5060]
28-Nov-2013 14:12:01.660   [CM503027]: Call(C:110): From: Extn:200 ("Baroni Limited" <sip:[email protected]:5060>)  to  T:Line:10005>>07770740133@[Dev:sip:[email protected]:5060]
28-Nov-2013 14:12:01.660   [CM503004]: Call(C:110): Route 3: from L:110.1[Extn] to T:Line:10005>>07770740133@[Dev:sip:[email protected]:5060]
28-Nov-2013 14:12:01.660   Line limit check: Current # of calls for line Lc:10005(@VoIPTalk[<sip:[email protected]:5060>]) is 0; limit is 2
28-Nov-2013 14:12:01.660   [Flow] Call(C:110): making call from L:110.1[Extn] to T:Line:10001>>07770740133@[Dev:sip:[email protected]:5062]
28-Nov-2013 14:12:01.660   [CM503027]: Call(C:110): From: Extn:200 ("Baroni Limited" <sip:[email protected]:5060>)  to  T:Line:10001>>07770740133@[Dev:sip:[email protected]:5062]
28-Nov-2013 14:12:01.660   [CM503004]: Call(C:110): Route 2: from L:110.1[Extn] to T:Line:10001>>07770740133@[Dev:sip:[email protected]:5062]
28-Nov-2013 14:12:01.660   Line limit check: Current # of calls for line Lc:10001(@OFFICE[<sip:[email protected]:5062>]) is 0; limit is 1
28-Nov-2013 14:12:01.660   [Flow] Call(C:110): making call from L:110.1[Extn] to T:Line:10002>>189907770740133@[Dev:sip:[email protected]:5062]
28-Nov-2013 14:12:01.660   [CM503027]: Call(C:110): From: Extn:200 ("Baroni Limited" <sip:[email protected]:5060>)  to  T:Line:10002>>189907770740133@[Dev:sip:[email protected]:5062]
28-Nov-2013 14:12:01.660   [CM503004]: Call(C:110): Route 1: from L:110.1[Extn] to T:Line:10002>>189907770740133@[Dev:sip:[email protected]:5062]
28-Nov-2013 14:12:01.660   Line limit check: Current # of calls for line Lc:10002(@HOME[<sip:[email protected]:5062>]) is 0; limit is 1
28-Nov-2013 14:12:01.660   Call(C:110): Call from Extn:200 to 189907770740133 matches outbound rule '1899'
28-Nov-2013 14:12:01.660   [Flow] Call(C:110): has built target endpoint: Out#:>>Rule{1899}>>189907770740133 for call from L:110.1[Extn]
28-Nov-2013 14:12:01.660   [Flow] Target endpoint for 189907770740133 is Out#:>>Rule{1899}>>189907770740133
28-Nov-2013 14:12:01.658   Caller '200' is found in the list of allowed extensions: 200-300
28-Nov-2013 14:12:01.658   Selected prefix: 1899
28-Nov-2013 14:12:01.658   Selected prefix: 1
28-Nov-2013 14:12:01.658   Selected prefix: 1
28-Nov-2013 14:12:01.658   Looking for outbound rule: dialed = [189907770740133], processed: [189907770740133]; from-ext:
28-Nov-2013 14:12:01.658   [Flow] Building target endpoint to 189907770740133 from "Baroni Limited" <sip:[email protected]:5060>
28-Nov-2013 14:12:01.658   [CM503010]: Call(C:110): Making route(s) from Extn:200 to <sip:[email protected]:5060>
28-Nov-2013 14:12:01.657   Remote SDP is set for leg L:110.1[Extn]
28-Nov-2013 14:12:01.657   OnOffer from "Baroni Limited"<sip:[email protected]>;tag=1328908937
28-Nov-2013 14:12:01.657   [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
28-Nov-2013 14:12:01.628   Inbound DID: ''; Phonebook Name: ''
28-Nov-2013 14:12:01.628   [CM500002]: Call(C:110): Info on incoming INVITE from Extn:200:
Invite-IN Recv Req INVITE from 192.168.1.90:5060 tid=2245037638dcc2ba22f67b44359c564d Call-ID=1786777608@192_168_1_90:
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK2245037638dcc2ba22f67b44359c564d;rport=5060
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected];user=phone>
From: "Baroni Limited"<sip:[email protected]>;tag=1328908937
Call-ID: 1786777608@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="414d535c08a7e03123:66e7eb247bec628884a7a016aa9772b2",response="8372e0c2f3125405b133a64800c0450a"
Supported: replaces
User-Agent: N300 IP/42.075.00.000.000
Allow-Events: message-summary, refer, ua-profile, talk
Content-Length: 379

v=0
o=200 5012 5 IN IP4 192.168.1.90
s=Mapping
c=IN IP4 192.168.1.90
t=0 0
m=audio 5012 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
28-Nov-2013 14:12:01.628   [CM503001]: Call(C:110): Incoming call from Extn:200 to <sip:[email protected]:5060>
28-Nov-2013 14:12:01.574   Outbound URI is used: sip:[email protected]:5060
28-Nov-2013 14:12:01.567   IncomingCall: C:110 from <sip:[email protected]:5060> to <sip:[email protected]:5060>
28-Nov-2013 14:12:01.561   Added leg L:C:110.1[No endpoint yet]
28-Nov-2013 14:12:01.538   UasSession 80837 started
28-Nov-2013 14:12:01.538   Call from "Baroni Limited"<sip:[email protected]>;tag=1328908937 to <sip:[email protected];user=phone>;tag=7a0ca106
28-Nov-2013 14:11:46.652   Line 10001 got Device: Dev(84):[sip:[email protected]:5062 / 10001]
28-Nov-2013 14:11:44.347   Currently active calls [none]
28-Nov-2013 14:11:14.093   Currently active calls [none]
28-Nov-2013 14:10:44.035   Currently active calls [none]
28-Nov-2013 14:10:14.000   Currently active calls [none]

Test 3:
Call made directly from handset through FXS (Line port on gateway) to PSTN via gateway FXO port:
This should NOT have anything to do with the 3CX, but clearly they are being redirected in to the 3CX instead of the FXO/PSTN... Must be routing table.

28-Nov-2013 14:08:13.754   Currently active calls [none]
28-Nov-2013 14:07:43.670   Currently active calls [none]
28-Nov-2013 14:07:36.504   L:109.3[Extn]: Terminating targets, reason: SIP ;cause=487 ;text="Request Cancelled"
28-Nov-2013 14:07:36.504   Leg L:109.3[Extn] is terminated: Cause: 487 Request Cancelled/INVITE from 192.168.1.91:5302
28-Nov-2013 14:07:36.504   L:109.3[Extn] got Terminated Recv 487/INVITE from 192.168.1.91:5302 tid=191e3149c000fa38 Call-ID=MDY5ZTQ5YzU1MjFmZTEwYzUxZjhjZTJlNTliMTk5OWE.:
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 192.168.1.51:5060;branch=z9hG4bK-d8754z-191e3149c000fa38-1---d8754z-;rport=5060
Contact: <sip:[email protected]:5302>
To: <sip:[email protected]>;tag=2953029285
From: "OFFICE"<sip:[email protected]:5060;nf=e>;tag=f13e0879
Call-ID: MDY5ZTQ5YzU1MjFmZTEwYzUxZjhjZTJlNTliMTk5OWE.
CSeq: 1 INVITE
User-Agent: N300A IP/42.075.00.000.000
Content-Length: 0
28-Nov-2013 14:07:36.504   Terminated from <sip:[email protected]>;tag=2953029285 to "OFFICE"<sip:[email protected]:5060;nf=e>;tag=f13e0879; reason: Error
28-Nov-2013 14:07:36.504   L:109.3[Extn] got Failure: Failure Recv 487/INVITE from 192.168.1.91:5302 tid=191e3149c000fa38 Call-ID=MDY5ZTQ5YzU1MjFmZTEwYzUxZjhjZTJlNTliMTk5OWE.:
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 192.168.1.51:5060;branch=z9hG4bK-d8754z-191e3149c000fa38-1---d8754z-;rport=5060
Contact: <sip:[email protected]:5302>
To: <sip:[email protected]>;tag=2953029285
From: "OFFICE"<sip:[email protected]:5060;nf=e>;tag=f13e0879
Call-ID: MDY5ZTQ5YzU1MjFmZTEwYzUxZjhjZTJlNTliMTk5OWE.
CSeq: 1 INVITE
User-Agent: N300A IP/42.075.00.000.000
Content-Length: 0
28-Nov-2013 14:07:36.504   [CM503003]: Call(C:109): Call to <sip:[email protected]:5060> has failed; Cause: 487 Request Cancelled/INVITE from 192.168.1.91:5302
28-Nov-2013 14:07:36.504   Session 80768 has failed in leg L:109.3[Extn] ; Cause: 487 Request Cancelled/INVITE from 192.168.1.91:5302
28-Nov-2013 14:07:36.504   Failure from <sip:[email protected]>;tag=2953029285 to "OFFICE"<sip:[email protected]:5060;nf=e>;tag=f13e0879
28-Nov-2013 14:07:36.462   [CM503008]: Call(C:109): Call is terminated
28-Nov-2013 14:07:36.459   L:109.3[Extn]: Terminating targets, reason: SIP ;cause=200 ;text="Call terminated on user request"
28-Nov-2013 14:07:36.423   Blocking refers for the Call(C:109)
28-Nov-2013 14:07:36.421   Call(C:109) is terminated
28-Nov-2013 14:07:36.413   ~Target=Ext:Ext.200
28-Nov-2013 14:07:36.411   ~Route=Dev:sip:[email protected]:5302
28-Nov-2013 14:07:36.402   Reason: SIP ;cause=487 ;text="Request Terminated"
28-Nov-2013 14:07:36.401   L:109.1[Line:10001<<10001]: Terminating targets, reason: SIP ;cause=487 ;text="Request Terminated"
28-Nov-2013 14:07:36.393   Leg L:109.1[Line:10001<<10001] is terminated: Cause: CANCEL from 192.168.1.95:5062
28-Nov-2013 14:07:36.371   L:109.1[Line:10001<<10001] got Terminated Recv Req CANCEL from 192.168.1.95:5062 tid=-4868e67b [email protected]:
CANCEL sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.95:5062;branch=z9hG4bK-4868e67b
Max-Forwards: 70
To: <sip:[email protected]>
From: "OFFICE"<sip:[email protected]>;tag=3c73e7795be70f83o0
Call-ID: [email protected]
CSeq: 102 CANCEL
Proxy-Authorization: Digest username="10001",realm="3CXPhoneSystem",nonce="414d535c08a7df1858:73fd3899dbdf72c1ac2d0f0a08556fbf",uri="sip:[email protected]",algorithm=MD5,response="2164921ff87f5bf55ed05aa1ac19a664"
User-Agent: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 0
28-Nov-2013 14:07:36.370   Terminated from "OFFICE"<sip:[email protected]>;tag=3c73e7795be70f83o0 to <sip:[email protected]>;tag=b256583a; reason: RemoteBye
28-Nov-2013 14:07:36.365   L:109.1[Line:10001<<10001] Sending: OnSendResp Send 487/INVITE from 0.0.0.0:0 tid=-4868e67b [email protected]:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.95:5062;branch=z9hG4bK-4868e67b
To: <sip:[email protected]>;tag=b256583a
From: "OFFICE"<sip:[email protected]>;tag=3c73e7795be70f83o0
Call-ID: [email protected]
CSeq: 102 INVITE
Content-Length: 0
28-Nov-2013 14:07:36.364   SendMsg from <sip:[email protected]>;tag=b256583a to "OFFICE"<sip:[email protected]>;tag=3c73e7795be70f83o0
28-Nov-2013 14:07:36.363   L:109.1[Line:10001<<10001] Sending: OnSendResp Send 200/CANCEL from 0.0.0.0:0 tid=-4868e67b [email protected]:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.95:5062;branch=z9hG4bK-4868e67b
Contact: <sip:[email protected]>
To: <sip:[email protected]>;tag=b256583a
From: "OFFICE"<sip:[email protected]>;tag=3c73e7795be70f83o0
Call-ID: [email protected]
CSeq: 102 CANCEL
Content-Length: 0
28-Nov-2013 14:07:36.360   SendMsg from <sip:[email protected]>;tag=b256583a to "OFFICE"<sip:[email protected]>;tag=3c73e7795be70f83o0
28-Nov-2013 14:07:29.060   L:109.2[Extn]: Terminating targets, reason: SIP ;cause=480 ;text="Temporarily not available"
28-Nov-2013 14:07:29.060   Leg L:109.2[Extn] is terminated: Cause: 480 Temporarily not available/INVITE from 192.168.1.90:5060
28-Nov-2013 14:07:29.060   L:109.2[Extn] got Terminated Recv 480/INVITE from 192.168.1.90:5060 tid=ff59281ad1744377 Call-ID=NWNmNDdjZTY3NDA1MDQ2ZDg0YTNiNTBhMDM4ZWU5NDg.:
SIP/2.0 480 Temporarily not available
Via: SIP/2.0/UDP 192.168.1.51:5060;branch=z9hG4bK-d8754z-ff59281ad1744377-1---d8754z-;rport=5060
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]>;tag=3056186545
From: "OFFICE"<sip:[email protected]:5060;nf=e>;tag=a27d1276
Call-ID: NWNmNDdjZTY3NDA1MDQ2ZDg0YTNiNTBhMDM4ZWU5NDg.
CSeq: 1 INVITE
User-Agent: N300 IP/42.075.00.000.000
Content-Length: 0
28-Nov-2013 14:07:29.060   Terminated from <sip:[email protected]>;tag=3056186545 to "OFFICE"<sip:[email protected]:5060;nf=e>;tag=a27d1276; reason: Error
28-Nov-2013 14:07:29.060   ~Route=Dev:sip:[email protected]:5060
28-Nov-2013 14:07:29.060   Call to T:Extn:200@[Dev:sip:[email protected]:5060,Dev:sip:[email protected]:5302] from L:109.1[Line:10001<<10001] failed, cause: Cause: 480 Temporarily not available/INVITE from 192.168.1.90:5060
28-Nov-2013 14:07:29.060   L:109.2[Extn] got Failure: Failure Recv 480/INVITE from 192.168.1.90:5060 tid=ff59281ad1744377 Call-ID=NWNmNDdjZTY3NDA1MDQ2ZDg0YTNiNTBhMDM4ZWU5NDg.:
SIP/2.0 480 Temporarily not available
Via: SIP/2.0/UDP 192.168.1.51:5060;branch=z9hG4bK-d8754z-ff59281ad1744377-1---d8754z-;rport=5060
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]>;tag=3056186545
From: "OFFICE"<sip:[email protected]:5060;nf=e>;tag=a27d1276
Call-ID: NWNmNDdjZTY3NDA1MDQ2ZDg0YTNiNTBhMDM4ZWU5NDg.
CSeq: 1 INVITE
User-Agent: N300 IP/42.075.00.000.000
Content-Length: 0
28-Nov-2013 14:07:29.060   [CM503003]: Call(C:109): Call to <sip:[email protected]:5060> has failed; Cause: 480 Temporarily not available/INVITE from 192.168.1.90:5060
28-Nov-2013 14:07:29.060   Session 80770 has failed in leg L:109.2[Extn] ; Cause: 480 Temporarily not available/INVITE from 192.168.1.90:5060
28-Nov-2013 14:07:29.060   Failure from <sip:[email protected]>;tag=3056186545 to "OFFICE"<sip:[email protected]:5060;nf=e>;tag=a27d1276
28-Nov-2013 14:07:21.093   L:109.1[Line:10001<<10001] Sending: OnSendResp Send 180/INVITE from 0.0.0.0:0 tid=-4868e67b [email protected]:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.95:5062;branch=z9hG4bK-4868e67b
Contact: <sip:[email protected]>
To: <sip:[email protected]>;tag=b256583a
From: "OFFICE"<sip:[email protected]>;tag=3c73e7795be70f83o0
Call-ID: [email protected]
CSeq: 102 INVITE
Content-Length: 0
28-Nov-2013 14:07:21.093   SendMsg from <sip:[email protected]>;tag=b256583a to "OFFICE"<sip:[email protected]>;tag=3c73e7795be70f83o0
28-Nov-2013 14:07:21.093   Provisional response arrived for session 80768 of Leg L:109.3[Extn]
28-Nov-2013 14:07:21.093   L:109.3[Extn] got Provisional Recv 180/INVITE from 192.168.1.91:5302 tid=191e3149c000fa38 Call-ID=MDY5ZTQ5YzU1MjFmZTEwYzUxZjhjZTJlNTliMTk5OWE.:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.51:5060;branch=z9hG4bK-d8754z-191e3149c000fa38-1---d8754z-;rport=5060
Contact: <sip:[email protected]:5302>
To: <sip:[email protected]>;tag=2953029285
From: "OFFICE"<sip:[email protected]:5060;nf=e>;tag=f13e0879
Call-ID: MDY5ZTQ5YzU1MjFmZTEwYzUxZjhjZTJlNTliMTk5OWE.
CSeq: 1 INVITE
User-Agent: N300A IP/42.075.00.000.000
Allow-Events: message-summary, refer, ua-profile, talk
Content-Length: 0
28-Nov-2013 14:07:21.093   Provisional(180) from <sip:[email protected]>;tag=2953029285 to "OFFICE"<sip:[email protected]:5060;nf=e>;tag=f13e0879
28-Nov-2013 14:07:21.093   [CM505001]: Endpoint Extn:200: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [N300A IP/42.075.00.000.000] PBX contact: [sip:[email protected]:5060]
28-Nov-2013 14:07:21.093   [CM503002]: Call(C:109): Alerting Extn:200 by contact <sip:[email protected]:5302>
28-Nov-2013 14:07:21.080   UacSession 80768 has formed leg L:109.3[Extn]
28-Nov-2013 14:07:21.080   Answer from <sip:[email protected]>;tag=2953029285 to "OFFICE"<sip:[email protected]:5060;nf=e>;tag=f13e0879
28-Nov-2013 14:07:20.967   [CM503025]: Call(C:109): Calling T:Extn:200@[Dev:sip:[email protected]:5060,Dev:sip:[email protected]:5302] for L:109.1[Line:10001<<10001]
28-Nov-2013 14:07:20.967   Route to L:109.3[Extn] sends Invite-OUT Send Req INVITE from 0.0.0.0:0 tid=fe033913ca431577 Call-ID=MDY5ZTQ5YzU1MjFmZTEwYzUxZjhjZTJlNTliMTk5OWE.:
INVITE sip:[email protected]:5302 SIP/2.0
Via: SIP/2.0/ ;branch=z9hG4bK-d8754z-fe033913ca431577-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]>
From: "OFFICE"<sip:[email protected]:5060;nf=e>;tag=f13e0879
Call-ID: MDY5ZTQ5YzU1MjFmZTEwYzUxZjhjZTJlNTliMTk5OWE.
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 419027746816 118027714561 IN IP4 192.168.1.51
s=3cxPS Audio call
c=IN IP4 192.168.1.51
t=0 0
m=audio 7022 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
28-Nov-2013 14:07:20.951   Outbound URI is used: sip:[email protected]:5302
28-Nov-2013 14:07:20.951   Added leg L:109.3[Extn]
28-Nov-2013 14:07:20.951   [CM503025]: Call(C:109): Calling T:Extn:200@[Dev:sip:[email protected]:5060,Dev:sip:[email protected]:5302] for L:109.1[Line:10001<<10001]
28-Nov-2013 14:07:20.951   Route to L:109.2[Extn] sends Invite-OUT Send Req INVITE from 0.0.0.0:0 tid=833e81089649d158 Call-ID=NWNmNDdjZTY3NDA1MDQ2ZDg0YTNiNTBhMDM4ZWU5NDg.:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/ ;branch=z9hG4bK-d8754z-833e81089649d158-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]>
From: "OFFICE"<sip:[email protected]:5060;nf=e>;tag=a27d1276
Call-ID: NWNmNDdjZTY3NDA1MDQ2ZDg0YTNiNTBhMDM4ZWU5NDg.
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 417937227776 290044510209 IN IP4 192.168.1.51
s=3cxPS Audio call
c=IN IP4 192.168.1.51
t=0 0
m=audio 7020 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
28-Nov-2013 14:07:20.926   Outbound URI is used: sip:[email protected]:5060
28-Nov-2013 14:07:20.911   Added leg L:109.2[Extn]
28-Nov-2013 14:07:20.873   [Flow] Call(C:109): making call from L:109.1[Line:10001<<10001] to T:Extn:200@[Dev:sip:[email protected]:5060,Dev:sip:[email protected]:5302]
28-Nov-2013 14:07:20.873   [CM503027]: Call(C:109): From: Line:10001<<10001 ("OFFICE" <sip:[email protected]:5060>)  to  T:Extn:200@[Dev:sip:[email protected]:5060,Dev:sip:[email protected]:5302]
28-Nov-2013 14:07:20.873   [CM503004]: Call(C:109): Route 1: from L:109.1[Line:10001<<10001] to T:Extn:200@[Dev:sip:[email protected]:5060,Dev:sip:[email protected]:5302]
28-Nov-2013 14:07:20.873   [Flow] Endpoint Extn:200 has no forwarding rule on reason 'All calls'
28-Nov-2013 14:07:20.873   [Flow] Call(C:109): has built target endpoint: Extn:200 for call from L:109.1[Line:10001<<10001]
28-Nov-2013 14:07:20.873   [Flow] Target endpoint for 200 is Extn:200
28-Nov-2013 14:07:20.873   [Flow] Building target endpoint to 200 from "OFFICE" <sip:[email protected]:5060>
28-Nov-2013 14:07:20.873   [CM503010]: Call(C:109): Making route(s) from Line:10001<<10001 to <sip:[email protected]:5060>
28-Nov-2013 14:07:20.871   Remote SDP is set for leg L:109.1[Line:10001<<10001]
28-Nov-2013 14:07:20.871   OnOffer from "OFFICE"<sip:[email protected]:5060;nf=e>
28-Nov-2013 14:07:20.871   [CM505002]: Gateway:[OFFICE] Device info: Device Identified: [Man: Linksys;Mod: SPA Series;Rev: General] Capabilities:[reinvite, no-replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA3102-5.2.13(GW002)] PBX contact: [sip:[email protected]:5060]
28-Nov-2013 14:07:20.846   Inbound DID: ''; Phonebook Name: ''
28-Nov-2013 14:07:20.846   [CM500002]: Call(C:109): Info on incoming INVITE from Line:10001<<10001:
Invite-IN Recv Req INVITE from 192.168.1.95:5062 tid=-4868e67b [email protected]:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.95:5062;branch=z9hG4bK-4868e67b
Max-Forwards: 70
Contact: "OFFICE"<sip:[email protected]:5062>
To: <sip:[email protected]>
From: "OFFICE"<sip:[email protected]>;tag=3c73e7795be70f83o0
Call-ID: [email protected]
CSeq: 102 INVITE
Expires: 240
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Content-Type: application/sdp
Proxy-Authorization: Digest username="10001",realm="3CXPhoneSystem",nonce="414d535c08a7df1858:73fd3899dbdf72c1ac2d0f0a08556fbf",uri="sip:[email protected]",algorithm=MD5,response="57bab5e315941acc440049c367408349"
Supported: x-sipura, replaces
User-Agent: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 441
Remote-Party-ID: OFFICE <sip:[email protected]>;screen=yes;party=calling

v=0
o=- 116265 116265 IN IP4 192.168.1.95
s=-
c=IN IP4 192.168.1.95
t=0 0
m=audio 9000 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
28-Nov-2013 14:07:20.846   [CM503001]: Call(C:109): Incoming call from Line:10001<<10001 to <sip:[email protected]:5060>
28-Nov-2013 14:07:20.828   Line limit check: Current # of calls for line Lc:10001(@OFFICE[<sip:[email protected]:5062>]) is 1; limit is 1
28-Nov-2013 14:07:20.828   Blacklist check: number '10001', list: ''; result = false
28-Nov-2013 14:07:20.808   Outbound URI is used: sip:[email protected]:5062
28-Nov-2013 14:07:20.794   [CM503012]: Inbound office hours rule (unnamed) for 10001 forwards to DN:200
28-Nov-2013 14:07:20.794   [Flow] Looking for inbound target: called=07770740133; caller="OFFICE" <sip:10001@:5060>
28-Nov-2013 14:07:20.794   CallerNameAddr: "OFFICE"<sip:10001;nf=e>
28-Nov-2013 14:07:20.794   Phonebook entry not found for number 10001
28-Nov-2013 14:07:20.794   Looking for phone number 10001 in tenant's 'default' phonebook
28-Nov-2013 14:07:20.794   No inbound caller ID reformat rule for DN:10001 is defined, or it is disabled ()
28-Nov-2013 14:07:20.794   Line 10001 has been updated from DB
28-Nov-2013 14:07:20.792   IncomingCall: C:109 from <sip:[email protected]:5060> to <sip:[email protected]:5060>
28-Nov-2013 14:07:20.792   Added leg L:C:109.1[No endpoint yet]
28-Nov-2013 14:07:20.791   UasSession 80764 started
28-Nov-2013 14:07:20.791   Call from "OFFICE"<sip:[email protected]>;tag=3c73e7795be70f83o0 to <sip:[email protected]>;tag=b256583a
28-Nov-2013 14:07:13.452   Currently active calls [none]
28-Nov-2013 14:06:42.953   Currently active calls [none]

 

[leejor]

The issue of my dial plan on the 3CX seems to have grown more complex. Just realised that ANY outbound call directed at going directly is now failing. I tried to make calls out from the FXS port on the gateway and all it did was call back in on the other gateway's inbound port account on the 3CX. B1oo6y confusing. I wonder if it is the Gateway dial plans. What does all this mean? It is the default settings. Could this be forcing the calls to fail.

(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)

I'm still puzzeled as to how you are using the FXS port. Did you register it as a 3Cx extension, if not why are you using it for outgoing calls directyly to the FXO port. I could understand putting your fax machine on it so that you can bypass having to convert to VoIP, but attempting to place phone calls out directly defeats the whole pint of having 3Cx in the first place.

As far as the dialplan goes... the default is a North American "catch all" plan that covers the basics. all calls will attempt to route out over the VoIP provider, if you want a call to go via the FXO port then that has to be specified.

Have a look at these sites for more information on crafting dialplans.

http://forum.voxilla.com/threads/how-do-i-craft-a-dial-plan-string.575/
http://www.netphonedirectory.com/pap2_dialplan2.htm
http://www.cisco.com/en/US/products/ps10033/products_qanda_item09186a0080a35a44.shtml

The 3102 is a bit different because of the GW0, forcing a call to the FXO port, and the ability to send calls to additional VoIP providers, but the rest is pretty much common to all Sipura/Linksys/Cisco devices.

When i get some time today, I'll look over the rest of your post.

 

[leejor]

I'm wondering if you have the 3102 set for two stage dialling. Without seeing all of the settings in the 3102, it appears that the digits you are sending out are being rejected by the 3102. this could be two stage dialling, or it is using one of it's internal dialplans and rejecting the number you are sending.

On the last part. You are sending digits to 3Cx from the 3102 (as a trunk) because you have not set-up the internal (Line) dialplan correctly in the 3102. Again, I'm not sure what yo are trying to do with the FXS port. Either disable it, or set it as a 3CX extension, if the power fails, the phone plugged into there will work for emergencies.

 

[Jonners59]

I'm wondering if you have the 3102 set for two stage dialling. Without seeing all of the settings in the 3102, it appears that the digits you are sending out are being rejected by the 3102. this could be two stage dialling, or it is using one of it's internal dialplans and rejecting the number you are sending.

Ok, that makes sense and where I was coming from, but I do not know what I am doing, so, how do I check for two stage dialling?

Dial plan, I do not, at this stage need any tampering by the 3102, I want the calls from the 3CX (VOIP) to go straight out to the FXO/PSTN, so what dial plan and where do I put it.

I note your comment about

(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)

, so what do i change for uK, better still do you know what each part means that I may alter it.

On the last part. You are sending digits to 3Cx from the 3102 (as a trunk) because you have not set-up the internal (Line) dialplan correctly in the 3102.

I guess this ties in with the above line in understanding the code for the dial plan. I just want to set it up as I had before such that IF my wife wants to just dial out via the PSTN without using the 3CX she can. On the handset we have it shows the extension off the 3CX which is the main way out and then also the direct line is also shown. My wife would like to be able to dial straight out if she wants to, and I can understand why. And what is wrong with the choice? Why limit the availability?

Again, I'm not sure what yo are trying to do with the FXS port. Either disable it, or set it as a 3CX extension, if the power fails, the phone plugged into there will work for emergencies.

Not what we want and have had for many years. Our setup is very simple and has planty of choice and auto failovers without complicating anything.

The 2 x PSTN lines are attached to 2 x FXO gateways (3102) and all inbound calls are routed via the VOIP port/account (10001 - WORK and 10002 - HOME) to Port virtual extensions on the 3CX, which in turn rout to one of two groups - Work (extn 200 to 204) or Home (205 to 299). 300 is for a fax machine. That is inbound.

Out calls are routed via different outbound rules, one for home and one for work to our sip provider, with alternative routes should the sip fail. The alternative routes differ depending if it is Home or Work, as all Work must be routed as an alternative via the Work Gateway/PSTN line. Again done using rules

Handsets. We use two Siemens N300 DECT SIP phones. They handle POTS PSTN as well as SIP. So the PSTN line is connected to the gateway FXO port and the DECT base station is also connected via an Ethernet cable to the LAN, thus I have set up two extensions on both base stations, one for Home (210) and One for Work (200) - both units are the same in VOIP account settings, but as each has one PSTN line connected, one for Home and One for Work in that they differ.

Forcing calls out over the PSTN from the 3CX, like doing it directly using the DECT phone option is to get access to local operator services and as my wife uses an account that works ONLY off the Home PSTN line to get cheaper calls she can route that way.... Either way, whether the call comes from the VOIP/3CX or the FXO it SHOULD just dial out, no digits added or stripped.

CLI issue.
Note the attached screenshot. Here a call is coming in and there is no CLI, what is showing is the account number on the 3CX. This is also what appears on the handset,. If I take a call straight off the PSTN line using the DECT the CLI is there.

 

[leejor]

Under the PSTN tab, VoIP to PSTN Gateway Setup/One stage dialling should be YES. In the PSTN to VoIp Gateway Setup/PSTN CID for VoIP CID should be YES.

As far as the dialplan for the FXS port, have a look over the sites I listed previously to understand the format. In the 3102, to send a call to the PSTN port (from the FXS) the string must have @GW0 following it.

For what you are truing to do, wouldn't it be easier to simply plug a phone into a spare socket if you wife doesn't want to call through 3CX?

You are going to have to craft a dialplan to accommodate any digits you might dial. You can go with a generic dialplan but then you have to wait for the timeout, or hit # to send the call.

 

[Jonners59]

Under the PSTN tab, VoIP to PSTN Gateway Setup/One stage dialling should be YES. In the PSTN to VoIp Gateway Setup/PSTN CID for VoIP CID should be YES.

Many thanks, these were already correctly set upon checking, so that can not be my issue....

As far as the dialplan for the FXS port, have a look over the sites I listed previously to understand the format. In the 3102, to send a call to the PSTN port (from the FXS) the string must have @GW0 following it.

From what I could see this is a dial plan...

(0-9xx.S0<:@gw0>|*xxS0<:@gw0>)

From what I can see it says any number starting 0 to9 with any number of digits after Straight Out Straight out via GW0 Then any number after * straight out and straight out over GW0, where GW0 is the FXS/PSTN port on the gateway.... Does this look correct to you? In other words I want ANYTHING typed to go out. No restrictions.

Then where do I put it and how do I use it?

For what you are truing to do, wouldn't it be easier to simply plug a phone into a spare socket if you wife doesn't want to call through 3CX?

YES, in a nutshell, however, it is tidier through the gateway AND more importantly, the when plugged in via the gateway the phone does not ring by itself, it ONLY rings via the 3CX, unless that fails, in which case the wonderful fail-over clicks in. SO ideally it stays... ALSO, to be honest over time I want my wife to try and use the 3CX for dialling out, and to only use the direct if the 3CX crashes, which has happened.

It all worked great before on the Grandstream and even the nasty Chinese box did this well. It WAS working on this box before I rebuilt it with the new configs, but because I do not understand it - yet - I do not know what it is that I have changed that caused it to stop working.

You are going to have to craft a dialplan to accommodate any digits you might dial. You can go with a generic dialplan but then you have to wait for the timeout, or hit # to send the call.

See above.....

Thanks for being patient with me Leejor

 

[leejor]

leejor wrote:For what you are truing to do, wouldn't it be easier to simply plug a phone into a spare socket if you wife doesn't want to call through 3CX?


YES, in a nutshell, however, it is tidier through the gateway AND more importantly, the when plugged in via the gateway the phone does not ring by itself, it ONLY rings via the 3CX, unless that fails, in which case the wonderful fail-over clicks in.

To do this (ring via 3CX), the Line (FXS port) must be registered as a 3CX extension, something that I was under the impression, you were not doing.

I'm not sure why CID isn't working, it could be that the "type" settings are not compatible with what BT is sending, or the delay, before that call is sent to 3CX is not long enough.

The dialplan that you create for the FXS port goes into the Line setting tab.

 

[Jonners59]

OK: CLI the FXO setting on the PSTN page should be 3 seconds. I recall this was also an issue on the Grandstream. Whilst the call info is sent in advance for some reason the gateways needed up to 3 seconds to collect/register the data. So that is now working. May be able to reduce that a bit with playing, but it is acceptable and works.

Now the direct outbound bit.

 

[leejor]

If the Line (FXS port) is not registered with a provider (3CX extension in this case), then 3CX has nothing to do with the set ringing. Is is probably a result of this option set to YES... PSTN Line/PSTN to VoIP Gateway Setup/ PSTN ring thru Line one

Which accomplishes the same thing as how you have your DECT sets analogue lines attached directly to the phone lines.

Normally the LINE 1 dialplan is used to direct calls to either the VoIP provider the line is registered to, or out on the PSTN line. If the line is not registered, or not enabled (not something I've tried), then I don't know how this is going to affect what happens.

About the simplest dialplan you can create, a sort of "catch all", would be (xx.<gw0>)

Normally, the 3102 will wait so many seconds after each digit (there is an timing option for this), to see if you have finished dialling. So after you last digit, you will have to wait, or try dialling #

This may, or may not handle all of the digit combinations that are dialled. If not, then you will have to learn the syntax of the dialplans and start experimenting. It's not difficult and besides the 3 links I provided, if you start searching, you will find many more, some, more specific to the SPA-3102, and it's older sibling, the SPA-3000.

As I said, for what you want to accomplish, it is much easier to just plug the set into a spare PSTN socket. You are going to a lot of unnecessary effort to come up with exactly the same result.

 

[Jonners59]

Thanks Leejor
The odd thing is that it WAS working earlier in the week, I just do not know what I changed to make it stop working, most infuriating.

I am certain it is to do with the FXO/PSTN accepting the call rather than before that point as the problem is from both the request to make a PSTN call from the 3CX and also from the FXS port, neither work. I have seriously been thinking of going back to scratch and using the a Oakley plan http://www.aoakley.com/articles/2008-01-08.php and then adding the 3CX as that is how I started.

Jonners

 

[leejor]

The odd thing is that it WAS working earlier in the week, I just do not know what I changed to make it stop working, most infuriating.

Thais is why it is important to document ANY changes you have made to the default settings, if you are not sure what you are doing. One seemingly simple change can drastically affect behaviour.

 

[Jonners59]

Tell me about it!!!