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

[Jonners59]

Gents Many thanks...

I made changes based on Chris's config to one of my devices. Some improvement. I can now send and receive calls to that device, but ONLY the attached phone. The calls are still NOT communicating with the 3CX. It is as if there was some sort of block not allowing the call through.

I have to go out in 30 mins for a few hours, shall be picking up later today. Please come back!!!

Best Jonners

 

[Jonners59]

OK, gents, before I go...

Added the settings to the other device and made some test calls. Both able to send calls from the attached handset directly out. So that is sorted.
If I use the 3CX to force the call to the gateway then the call is sent to the gateway, but does not seem to make the call from the gateway to the PSTN - I can see the call in the status.

Calls arriving at the gateway are received and answered by the gateway but are just failing to connect to the 3CX or send to the attached handset. In active service I only want the handset to work if there is no power or the device is not registered to the 3CX for some reason.. i.e. to use in the event of emergency as a fallback.

Best Jonners

 

[Cjay]

Is there life after 3CX? In case you are anxious to spend more time prodding and poking here are some more thoughts:

You mention:

even better for UK peeps

however to my way of thinking this site is intended to set up the 3102 as a standalone ATA/Voip device without using anything else (like lovely shiny 3CX). So whilst the UK dial tones etc are correct the rest of the setup is is not intended to make it function as a gateway for 3CX. I suspect from your last comments you may be using it in this mode.

I would suggest going back to first principles:
Ensure 3102 is connected to rest of your network via the Internet (blue) port
Do not plug an analog phone into the 3102 phone port (at least not until it is all working as intended), test using a separate IP phone through 3CX.
Ensure the 3102 firmware is reasonably current - mine is 5.1.10(GW)
Turn off firewall on your 3cx computer while you set everything up - it might get in the way!

If you have the patience, set up your 3102 exactly like the config I sent you - except for:
Line 1: Set Line Enable to No and ignore the rest of the Line 1 settings.
PSTN:
- SIP port must match what you have set in 3cx gateway setting
- Proxy is your 3cx IP
- Password/Auth ID must match your 3cx settings

As Leejor suggested, remove the DDI in 3cx. Simply edit the 3cx VOIP/PSTN gateway / BT line / Port 1000x / Port details to route incoming calls to a valid internal extension. You can subsequently change this to a ring group or other fancieness when all is working ok!

Ensure you have a valid outbound rule to route IP dialled numbers out over your BT PSTN line.

Power cycle 3102.

Does the port/trunks status show the device as registered? What happens if you call in/out?

Hope some of this helps...

Regds,
Chris

 

[Jonners59]

Chris
Many thanks... Update:

I would suggest going back to first principles:
Ensure 3102 is connected to rest of your network via the Internet (blue) port
Do not plug an analog phone into the 3102 phone port (at least not until it is all working as intended), test using a separate IP phone through 3CX.
Ensure the 3102 firmware is reasonably current - mine is 5.1.10(GW)
Turn off firewall on your 3cx computer while you set everything up - it might get in the way!

Already done, only used the blue LAN Ethernet port.
Unplugged the phone for now.
Up to date on firmware, ahead of yours which made matching a little tricky as lines did not match up.
Not turned off firewall as yet.

If you have the patience, set up your 3102 exactly like the config I sent you - except for:
Line 1: Set Line Enable to No and ignore the rest of the Line 1 settings.
PSTN:
- SIP port must match what you have set in 3cx gateway setting
- Proxy is your 3cx IP
- Password/Auth ID must match your 3cx settings

Matched yours exactly, bar the account settings I am using

As Leejor suggested, remove the DDI in 3cx. Simply edit the 3cx VOIP/PSTN gateway / BT line / Port 1000x / Port details to route incoming calls to a valid internal extension. You can subsequently change this to a ring group or other fancieness when all is working ok!

Already done

Ensure you have a valid outbound rule to route IP dialled numbers out over your BT PSTN line.

Already done, and worked before... Also during test calls I could see them trying as they should so no issues there.

Power cycle 3102.

Done

Does the port/trunks status show the device as registered? What happens if you call in/out?

On the 3CX they have always shown as registered.... No calls, still fail in and out.

Outbound call log below, FYI:

24-Nov-2013 19:47:37.194   L:5.2[Line:10001>>02086689100] Sending: OnSendReq Send Req ACK from 0.0.0.0:0 tid=5e06ff0f5600590f Call-ID=ZTJlMmU0MGE0NzEyODdkNjFkOTQ2ZDhjYWU4NTUyNDc.:
ACK sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/ ;branch=z9hG4bK-d8754z-5e06ff0f5600590f-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]:5062>;tag=c16f51928bd3437ei1
From: "Baroni Limited"<sip:[email protected]:5060>;tag=9210f765
Call-ID: ZTJlMmU0MGE0NzEyODdkNjFkOTQ2ZDhjYWU4NTUyNDc.
CSeq: 1 ACK
User-Agent: 3CXPhoneSystem 12.0.32816.397 (32731)
Content-Length: 0
24-Nov-2013 19:47:37.194   SendMsg from "Baroni Limited"<sip:[email protected]:5060>;tag=9210f765 to <sip:[email protected]:5062>;tag=c16f51928bd3437ei1
24-Nov-2013 19:47:33.634   L:5.2[Line:10001>>02086689100] Sending: OnSendReq Send Req ACK from 0.0.0.0:0 tid=5e06ff0f5600590f Call-ID=ZTJlMmU0MGE0NzEyODdkNjFkOTQ2ZDhjYWU4NTUyNDc.:
ACK sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/ ;branch=z9hG4bK-d8754z-5e06ff0f5600590f-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]:5062>;tag=c16f51928bd3437ei1
From: "Baroni Limited"<sip:[email protected]:5060>;tag=9210f765
Call-ID: ZTJlMmU0MGE0NzEyODdkNjFkOTQ2ZDhjYWU4NTUyNDc.
CSeq: 1 ACK
User-Agent: 3CXPhoneSystem 12.0.32816.397 (32731)
Content-Length: 0
24-Nov-2013 19:47:33.634   SendMsg from "Baroni Limited"<sip:[email protected]:5060>;tag=9210f765 to <sip:[email protected]:5062>;tag=c16f51928bd3437ei1
24-Nov-2013 19:47:31.853   L:5.2[Line:10001>>02086689100] Sending: OnSendReq Send Req ACK from 0.0.0.0:0 tid=5e06ff0f5600590f Call-ID=ZTJlMmU0MGE0NzEyODdkNjFkOTQ2ZDhjYWU4NTUyNDc.:
ACK sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/ ;branch=z9hG4bK-d8754z-5e06ff0f5600590f-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]:5062>;tag=c16f51928bd3437ei1
From: "Baroni Limited"<sip:[email protected]:5060>;tag=9210f765
Call-ID: ZTJlMmU0MGE0NzEyODdkNjFkOTQ2ZDhjYWU4NTUyNDc.
CSeq: 1 ACK
User-Agent: 3CXPhoneSystem 12.0.32816.397 (32731)
Content-Length: 0
24-Nov-2013 19:47:31.853   SendMsg from "Baroni Limited"<sip:[email protected]:5060>;tag=9210f765 to <sip:[email protected]:5062>;tag=c16f51928bd3437ei1
24-Nov-2013 19:47:30.959   L:5.2[Line:10001>>02086689100] Sending: OnSendReq Send Req ACK from 0.0.0.0:0 tid=5e06ff0f5600590f Call-ID=ZTJlMmU0MGE0NzEyODdkNjFkOTQ2ZDhjYWU4NTUyNDc.:
ACK sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/ ;branch=z9hG4bK-d8754z-5e06ff0f5600590f-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]:5062>;tag=c16f51928bd3437ei1
From: "Baroni Limited"<sip:[email protected]:5060>;tag=9210f765
Call-ID: ZTJlMmU0MGE0NzEyODdkNjFkOTQ2ZDhjYWU4NTUyNDc.
CSeq: 1 ACK
User-Agent: 3CXPhoneSystem 12.0.32816.397 (32731)
Content-Length: 0
24-Nov-2013 19:47:30.959   SendMsg from "Baroni Limited"<sip:[email protected]:5060>;tag=9210f765 to <sip:[email protected]:5062>;tag=c16f51928bd3437ei1
24-Nov-2013 19:47:30.754   Session 1158 of leg L:5.1[Extn] is confirmed
24-Nov-2013 19:47:30.748   L:5.1[Extn] got Confirmed Recv Req ACK from 192.168.1.90:5060 tid=1f73056520aececfc0509514da82e852 Call-ID=1849662081@192_168_1_90:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK1f73056520aececfc0509514da82e852;rport=5060
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected];user=phone>;tag=93612540
From: "Baroni Limited"<sip:[email protected]>;tag=2592517405
Call-ID: 1849662081@192_168_1_90
CSeq: 3 ACK
Proxy-Authorization: Digest username="200", realm="3CXPhoneSystem", algorithm=MD5, uri="sip:[email protected];user=phone", nonce="414d535c08a2e8d231:640e9c6393b114d6c23e802cb0b8ab12", response="598432c7a64fd638a4f59cc9fd890148"
User-Agent: N300 IP/42.075.00.000.000
Content-Length: 0
24-Nov-2013 19:47:30.739   ConnCfmd from "Baroni Limited"<sip:[email protected]>;tag=2592517405 to <sip:[email protected];user=phone>;tag=93612540
24-Nov-2013 19:47:30.579   [CM503007]: Call(C:5): Line:10001>>02086689100 has joined, contact <sip:[email protected]:5062>
24-Nov-2013 19:47:30.551   [MS004000] C:5.2: (ACTIVE): Can't send RTP stream to 127.0.0.1:9008 destination unreachable
24-Nov-2013 19:47:30.551   MSLogReports
24-Nov-2013 19:47:30.549   [CM503007]: Call(C:5): Extn:200 has joined, contact <sip:[email protected]:5060>
24-Nov-2013 19:47:30.549   ~Target=PSTNline:02086689100@(Ln.10001@OFFICE)
24-Nov-2013 19:47:30.549   Session 1158 of leg L:5.1[Extn] is connected
24-Nov-2013 19:47:30.549   L:5.1[Extn] got Connected.UAS Send 200/INVITE from 0.0.0.0:0 tid=d910a7ca581f409d2d1a1f6ccca4110 Call-ID=1849662081@192_168_1_90:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bKd910a7ca581f409d2d1a1f6ccca4110;rport=5060
Contact: <sip:[email protected]:5060>
To: <sip:[email protected];user=phone>;tag=93612540
From: "Baroni Limited"<sip:[email protected]>;tag=2592517405
Call-ID: 1849662081@192_168_1_90
CSeq: 3 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhoneSystem 12.0.32816.397 (32731)
Content-Length: 269

v=0
o=3cxPS 8640266240 98331262977 IN IP4 192.168.1.51
s=3cxPS Audio call
c=IN IP4 192.168.1.51
t=0 0
m=audio 7010 RTP/AVP 9 8 0 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
24-Nov-2013 19:47:30.549   Connected(UAS) from "Baroni Limited"<sip:[email protected]>;tag=2592517405 to <sip:[email protected];user=phone>;tag=93612540
24-Nov-2013 19:47:30.549   L:5.1[Extn] Sending: OnSendResp Send 200/INVITE from 0.0.0.0:0 tid=d910a7ca581f409d2d1a1f6ccca4110 Call-ID=1849662081@192_168_1_90:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bKd910a7ca581f409d2d1a1f6ccca4110;rport=5060
Contact: <sip:[email protected]:5060>
To: <sip:[email protected];user=phone>;tag=93612540
From: "Baroni Limited"<sip:[email protected]>;tag=2592517405
Call-ID: 1849662081@192_168_1_90
CSeq: 3 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
Content-Length: 269

v=0
o=3cxPS 8640266240 98331262977 IN IP4 192.168.1.51
s=3cxPS Audio call
c=IN IP4 192.168.1.51
t=0 0
m=audio 7010 RTP/AVP 9 8 0 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
24-Nov-2013 19:47:30.549   SendMsg from <sip:[email protected];user=phone>;tag=93612540 to "Baroni Limited"<sip:[email protected]>;tag=2592517405
24-Nov-2013 19:47:30.533   ~Target=PSTNline:02086689100@(Ln.10002@HOME)
24-Nov-2013 19:47:30.533   ~Route=Dev:sip:[email protected]:5062
24-Nov-2013 19:47:30.533   ~Target=VoIPline:02086689100@(Ln.10005@VoIPTalk)
24-Nov-2013 19:47:30.533   ~Route=Dev:sip:[email protected]:5060
24-Nov-2013 19:47:30.533   L:5.1[Extn]: Terminating targets, reason: SIP ;cause=200 ;text="Call completed elsewhere"
24-Nov-2013 19:47:30.528   ~Route=Dev:sip:[email protected]:5062
24-Nov-2013 19:47:30.528   L:5.2[Line:10001>>02086689100] has joined to L:5.1[Extn]
24-Nov-2013 19:47:30.528   Session 1161 of leg L:5.2[Line:10001>>02086689100] is connected
24-Nov-2013 19:47:30.528   L:5.2[Line:10001>>02086689100] got Connected.UAC Recv 200/INVITE from 192.168.1.95:5062 tid=e2370f13ce487422 Call-ID=ZTJlMmU0MGE0NzEyODdkNjFkOTQ2ZDhjYWU4NTUyNDc.:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.51:5060;branch=z9hG4bK-d8754z-e2370f13ce487422-1---d8754z-
Contact: "HOME"<sip:[email protected]:5062>
To: <sip:[email protected]:5062>;tag=c16f51928bd3437ei1
From: "Baroni Limited"<sip:[email protected]:5060>;tag=9210f765
Call-ID: ZTJlMmU0MGE0NzEyODdkNjFkOTQ2ZDhjYWU4NTUyNDc.
CSeq: 1 INVITE
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Content-Type: application/sdp
Server: Linksys/SPA3102-5.2.13(GW002)
Supported: x-sipura, replaces
Content-Length: 244
Remote-Party-ID: HOME <sip:[email protected]>;screen=yes;party=called

v=0
o=- 34153 34153 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=audio 9008 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/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
24-Nov-2013 19:47:30.528   Connected(UAC) from <sip:[email protected]:5062>;tag=c16f51928bd3437ei1 to "Baroni Limited"<sip:[email protected]:5060>;tag=9210f765
24-Nov-2013 19:47:30.528   Remote SDP is set for leg L:5.2[Line:10001>>02086689100]
24-Nov-2013 19:47:30.528   Answer SDP arrived on Leg L:5.2[Line:10001>>02086689100];
OnAnswer Recv 200/INVITE from 192.168.1.95:5062 tid=e2370f13ce487422 Call-ID=ZTJlMmU0MGE0NzEyODdkNjFkOTQ2ZDhjYWU4NTUyNDc.:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.51:5060;branch=z9hG4bK-d8754z-e2370f13ce487422-1---d8754z-
Contact: "HOME"<sip:[email protected]:5062>
To: <sip:[email protected]:5062>;tag=c16f51928bd3437ei1
From: "Baroni Limited"<sip:[email protected]:5060>;tag=9210f765
Call-ID: ZTJlMmU0MGE0NzEyODdkNjFkOTQ2ZDhjYWU4NTUyNDc.
CSeq: 1 INVITE
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Content-Type: application/sdp
Server: Linksys/SPA3102-5.2.13(GW002)
Supported: x-sipura, replaces
Content-Length: 244
Remote-Party-ID: HOME <sip:[email protected]>;screen=yes;party=called

v=0
o=- 34153 34153 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=audio 9008 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/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
24-Nov-2013 19:47:30.528   [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]
24-Nov-2013 19:47:30.528   [CM503002]: Call(C:5): Alerting Line:10001>>02086689100 by contact <sip:[email protected]:5062>
24-Nov-2013 19:47:30.515   Phonebook entry not found for number 02086689100
24-Nov-2013 19:47:30.515   Looking for phone number 02086689100 in tenant's 'default' phonebook
24-Nov-2013 19:47:30.515   UacSession 1161 has formed leg L:5.2[Line:10001>>02086689100]
24-Nov-2013 19:47:30.515   Answer from <sip:[email protected]:5062>;tag=c16f51928bd3437ei1 to "Baroni Limited"<sip:[email protected]:5060>;tag=9210f765
24-Nov-2013 19:47:30.515   L:5.2[Line:10001>>02086689100] Sending: OnSendReq Send Req ACK from 0.0.0.0:0 tid=5e06ff0f5600590f Call-ID=ZTJlMmU0MGE0NzEyODdkNjFkOTQ2ZDhjYWU4NTUyNDc.:
ACK sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/ ;branch=z9hG4bK-d8754z-5e06ff0f5600590f-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]:5062>;tag=c16f51928bd3437ei1
From: "Baroni Limited"<sip:[email protected]:5060>;tag=9210f765
Call-ID: ZTJlMmU0MGE0NzEyODdkNjFkOTQ2ZDhjYWU4NTUyNDc.
CSeq: 1 ACK
Content-Length: 0
24-Nov-2013 19:47:30.515   SendMsg from "Baroni Limited"<sip:[email protected]:5060>;tag=9210f765 to <sip:[email protected]:5062>;tag=c16f51928bd3437ei1
24-Nov-2013 19:47:30.438   [CM503025]: Call(C:5): Calling T:Line:10001>>02086689100@[Dev:sip:[email protected]:5062] for L:5.1[Extn]
24-Nov-2013 19:47:30.438   Route to L:5.2[Line:10001>>02086689100] sends Invite-OUT Send Req INVITE from 0.0.0.0:0 tid=df4f784ae776944b Call-ID=ZTJlMmU0MGE0NzEyODdkNjFkOTQ2ZDhjYWU4NTUyNDc.:
INVITE sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/ ;branch=z9hG4bK-d8754z-df4f784ae776944b-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]:5062>
From: "Baroni Limited"<sip:[email protected]:5060>;tag=9210f765
Call-ID: ZTJlMmU0MGE0NzEyODdkNjFkOTQ2ZDhjYWU4NTUyNDc.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
Content-Length: 356

v=0
o=3cxPS 13572767744 148847460353 IN IP4 192.168.1.51
s=3cxPS Audio call
c=IN IP4 192.168.1.51
t=0 0
m=audio 7012 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
24-Nov-2013 19:47:30.406   Outbound URI is used: sip:[email protected]:5062
24-Nov-2013 19:47:30.400   Added leg L:5.2[Line:10001>>02086689100]
24-Nov-2013 19:47:30.375   [Flow] Call(C:5): making call from L:5.1[Extn] to T:Line:10002>>02086689100@[Dev:sip:[email protected]:5062]
24-Nov-2013 19:47:30.375   [CM503027]: Call(C:5): From: Extn:200 ("Baroni Limited" <sip:[email protected]:5060>)  to  T:Line:10002>>02086689100@[Dev:sip:[email protected]:5062]
24-Nov-2013 19:47:30.375   [CM503004]: Call(C:5): Route 3: from L:5.1[Extn] to T:Line:10002>>02086689100@[Dev:sip:[email protected]:5062]
24-Nov-2013 19:47:30.375   Line limit check: Current # of calls for line Lc:10002(@HOME[<sip:[email protected]:5062>]) is 0; limit is 1
24-Nov-2013 19:47:30.375   [Flow] Call(C:5): making call from L:5.1[Extn] to T:Line:10005>>02086689100@[Dev:sip:[email protected]:5060]
24-Nov-2013 19:47:30.375   [CM503027]: Call(C:5): From: Extn:200 ("Baroni Limited" <sip:[email protected]:5060>)  to  T:Line:10005>>02086689100@[Dev:sip:[email protected]:5060]
24-Nov-2013 19:47:30.375   [CM503004]: Call(C:5): Route 2: from L:5.1[Extn] to T:Line:10005>>02086689100@[Dev:sip:[email protected]:5060]
24-Nov-2013 19:47:30.375   Line limit check: Current # of calls for line Lc:10005(@VoIPTalk[<sip:[email protected]:5060>]) is 0; limit is 2
24-Nov-2013 19:47:30.375   [Flow] Call(C:5): making call from L:5.1[Extn] to T:Line:10001>>02086689100@[Dev:sip:[email protected]:5062]
24-Nov-2013 19:47:30.375   [CM503027]: Call(C:5): From: Extn:200 ("Baroni Limited" <sip:[email protected]:5060>)  to  T:Line:10001>>02086689100@[Dev:sip:[email protected]:5062]
24-Nov-2013 19:47:30.375   [CM503004]: Call(C:5): Route 1: from L:5.1[Extn] to T:Line:10001>>02086689100@[Dev:sip:[email protected]:5062]
24-Nov-2013 19:47:30.375   Line limit check: Current # of calls for line Lc:10001(@OFFICE[<sip:[email protected]:5062>]) is 0; limit is 1
24-Nov-2013 19:47:30.375   Call(C:5): Call from Extn:200 to 402086689100 matches outbound rule 'PSTN Baroni'
24-Nov-2013 19:47:30.375   [Flow] Call(C:5): has built target endpoint: Out#:>>Rule{PSTN Baroni}>>402086689100 for call from L:5.1[Extn]
24-Nov-2013 19:47:30.375   [Flow] Target endpoint for 402086689100 is Out#:>>Rule{PSTN Baroni}>>402086689100
24-Nov-2013 19:47:30.375   Selected prefix: 4
24-Nov-2013 19:47:30.375   Looking for outbound rule: dialed = [402086689100], processed: [402086689100]; from-ext:
24-Nov-2013 19:47:30.375   [Flow] Building target endpoint to 402086689100 from "Baroni Limited" <sip:[email protected]:5060>
24-Nov-2013 19:47:30.375   [CM503010]: Call(C:5): Making route(s) from Extn:200 to <sip:[email protected]:5060>
24-Nov-2013 19:47:30.375   Remote SDP is set for leg L:5.1[Extn]
24-Nov-2013 19:47:30.375   OnOffer from "Baroni Limited"<sip:[email protected]>;tag=2592517405
24-Nov-2013 19:47:30.375   [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]
24-Nov-2013 19:47:30.354   Inbound DID: ''; Phonebook Name: ''
24-Nov-2013 19:47:30.354   [CM500002]: Call(C:5): Info on incoming INVITE from Extn:200:
Invite-IN Recv Req INVITE from 192.168.1.90:5060 tid=d910a7ca581f409d2d1a1f6ccca4110 Call-ID=1849662081@192_168_1_90:
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bKd910a7ca581f409d2d1a1f6ccca4110;rport=5060
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected];user=phone>
From: "Baroni Limited"<sip:[email protected]>;tag=2592517405
Call-ID: 1849662081@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="414d535c08a2e8d231:640e9c6393b114d6c23e802cb0b8ab12",response="598432c7a64fd638a4f59cc9fd890148"
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 5012 227 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
24-Nov-2013 19:47:30.354   [CM503001]: Call(C:5): Incoming call from Extn:200 to <sip:[email protected]:5060>

 

[leejor]

Some improvement. I can now send and receive calls to that device, but ONLY the attached phone.

As has been mentioned, don't plug a phone into the 3102, stating that it is able to place calls has no bearing on the 3102 working as a gateway to 3CX. Later, you can plug a phone in there, if you wish, but, as an emergency set, you'd do just as well plugging it into any spare socket in the house and turning down the ringer. The FXS port is better utilized as an extension off 3CX, set-up AFTER you get the FXO port working properly. If the FXS is used as an extension, then a power failure will connect that set directly to the PSTN line, until then it can use 3CX to route calls, which is whole purpose of installing 3CX in the first place, isn't it?.

If you reach a point that you have pulled out all of your hair, do a full factory reset on the 3102 (that's the dial r.e.s.e.t method). Delete your trunk in 3CX, and start over. Sometimes one small change, that was forgotten about, can throw the whole thing off. As mentioned, be sure that you are using the Blue Ethernet socket to connect to your network. If you follow the guide that I posted the link for, you should at least, be able to make and receive calls. All of the other, UK specific changes, can be made later, but make notes, as you change settings.

 

[leejor]

Eliminate your VoIP provider, as the second choice route for your test calls, it seems to be going out that route but there is no appearance reason why.

Going by the IP's, you seem to have some 192.168.1.X and 192.168.100.X...I hope that all your devices have their subnet set to 255.255.0.0 or they won't be able to communicate. Did you make a typo when putting in some IP addresses?

 

[Jonners59]

Leejor
I have a choice as to where the calls are routed out.
As the gateway is not accepting calls sent directly from the sip phone to set up inbound tests I am then either calling via the VoIP provider or allowing the call to over flow to the back up route, the VoIP provider. The gateway and 3CX are just not talking.

Re the IP address, that is odd. I set the gateway's WAN to 192.168.100.95 and 192.168.100.96 so that it did not clash with the LAN port 192.168.1.95 and 192.168.1.96, as I read that they should be different as otherwise it causes problems with the 3CX.

I'll try again with the IP addresses the same.

I have to say fine tuning with the FW upgrade and Chris's settings the gateway is now not even accepting inbound calls, which it was before, which brings me to, how do i back up the device?

 

[Jonners59]

changed it, now can't log on!

 

[Cjay]

Ouch, can you give us an overview of the IP addressing scheme you are using on your network?

 

[leejor]

Re the IP address, that is odd. I set the gateway's WAN to 192.168.100.95 and 192.168.100.96 so that it did not clash with the LAN port 192.168.1.95 and 192.168.1.96, as I read that they should be different as otherwise it causes problems with the 3CX.

You should not even be using the router within the 3102. You first have to change the 3102 option to allow access from the WAN side. Under the WAN setup tab, Remote Management/Enable WAN Web Server: should be set to Yes If you are unable to log-in , plug in a set, dial **** then 7932#, that allows you to toggle this setting on and off.
You should not be changing any settings under the LAN setup tab unless you plan on using the built in router, most don't. In fact, once the 3102 is up and running properly, the Networking Service should be set to Bridge, and disable DHCP server.

The logs you posted show the use of 192.168.100.X IP's

Line:10001>>02086689100 has joined, contact <sip:[email protected]:5062>

That is a different subnet and may be what is causing the problems.

The WAN setup tab is what you should be concerned about. The IP used there (same subnet as the rest of your network), is the same IP that you should have in the 3CX settings that you created for that trunk. That is how 3CX sends a call to the gateway.

 

[Jonners59]

Ahhhhh... I was aware of the built in router and how that can cause issues, and have been using the ethernet. I did set bridge yesterday but I could not get in after so reset the device. OK. Can you send what you have done on yours for ref? I have to rebuild one of the gateways from scratch.... lets see, PS: IP addresses below

 

[Cjay]

Emailed router part to you. Do make sure your LAN cable is plugged into blue Internet port - not Ethernet!
Chris

 

[leejor]

From a factory "fresh" start, you first log-in using a computer plugged into the Yellow (Ethernet) port. that computer will get an IP from the built in router. First thing is to enable the WAN web server (as mentioned above) this gives you access to the device from the WAN (blue) socket, the one you should use AFTER you give the 3102 a fixed IP in the WAN settings.

Don't enable Bridge, on the LAN side, or disable DHCP until after you are sure that you can access the 3102 from the WAN side. No other settings on the LAN side should need to be changed unless there is a conflict with the default IP of the 3102 (LAN IP address). Since the default is 192.168.0.1, it should not make a difference on your network.

On the WAN side, you set a static IP, the gateway is the IP of your router (same one as in the 3CX trunk settings), and you can set the DNS server as your router IP, or a public DNS server IP. You can set an NTP server if you want, it's not critical, but if the time zone is set correctly, then you'll know if the 3102 is able to get out to the internet because the clock will be correct. This is also necessary if you connect a 3CX extension to the FXS port or caller ID time/date will be incorrect.

 

[Jonners59]

From a factory "fresh" start, you first log-in using a computer plugged into the Yellow (Ethernet) port. that computer will get an IP from the built in router. First thing is to enable the WAN web server (as mentioned above) this gives you access to the device from the WAN (blue) socket, the one you should use AFTER you give the 3102 a fixed IP in the WAN settings.

Don't enable Bridge, on the LAN side, or disable DHCP until after you are sure that you can access the 3102 from the WAN side. No other settings on the LAN side should need to be changed unless there is a conflict with the default IP of the 3102 (LAN IP address). Since the default is 192.168.0.1, it should not make a difference on your network.

On the WAN side, you set a static IP, the gateway is the IP of your router (same one as in the 3CX trunk settings), and you can set the DNS server as your router IP, or a public DNS server IP. You can set an NTP server if you want, it's not critical, but if the time zone is set correctly, then you'll know if the 3102 is able to get out to the internet because the clock will be correct. This is also necessary if you connect a 3CX extension to the FXS port or caller ID time/date will be incorrect.

Chris
many thanks for the files...

Leejor
All as above done and dusted, and the install is identical to that of Chris, bar account settings and my SW is a higher version. I have also now made the changes to the connect using the WAN port..... Still no joy.
Seem to be going backwards. Before I was able to get inbound calls to register on the gateway, now they do not do that and one account does not register on the 3CX... yet settings are the same!

 

[Jonners59]

OK, seems that I have it working...... I need to run some more tests as it is getting late I have not done all that I need to do, and I do have an issue with the inbound calls not going where they should, may be due to deleting the DID or at least I can not see as used to be in the v9 3CX where inbound calls route....

So that's it for tonight. I shall try and resume in the next evening or two and get back on tests....

Guys many thanks so far

PS> How do i do a back up?

 

[leejor]

This explains how to back-up a Cisco/Linksys SPA device.

https://supportforums.cisco.com/docs/DOC-9896

A printed copy, of each Tab, highlighting the fields that you have changed from default settings, can also be helpful in understanding what is going on.

 

[Jonners59]

OK guys
It is sort of working:
I have also done the backup, thanks for the link. Now understand how you managed to send those pges without a zillion screen-shots

The issues I have left seem to be:

• Inbound calls do not have CLI (Calling Line Identification). All I get is the Account name of the gateway port the call is coming in on on the 3CX
  
  I get echo at the 3CX end. The external party the sound is a little too loud, I assume that is a gain setting somewhere.
  
  If the Ethernet cable is disconnected or I change the account settings so the gateway does not register - simulating lost 3CX, the gateway does not go in to fail-over, i.e. route the call to the FXS port and the POT.
  
  Dial Plans: I have a couple of dial plans on the 3CX to route calls starting with 1 (one) to the FXO port. They used to work, but now they do not work. The call does not go out, and ends up being routed via the VoIP service provider instead - 3rd option. See attached

Jonners

 

[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.

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.

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.

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.

 

[amygoda]

Hello,
Update firmware to 5.1.7
then reset to factory set

 

[leejor]

Yes, it is a good idea to upgrade the firmware as later releases seem to have fixed some of the echo issues, although having the correct settings is of much more help.

See...

http://www.cisco.com/en/US/products/ps10024/products_qanda_item09186a0080a35f1e.shtml

and

http://software.cisco.com/download/release.html?mdfid=282414112&softwareid=282463187&release=5.2.13&relind=AVAILABLE&rellifecycle=&reltype=latest

Have a look at this as it might affect you (you may want to do a search for more info on the web).

https://supportforums.cisco.com/thread/2148348

A "factory reset isn't necessary after a firmware upgrade, all of your settings are retained, if it has been a while since you did one, and can't remember all of the settings you have changed, then one might be needed. If you do, you might want to have (print out) a copy of the setting you are currently using.