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

Discussion in '3CX Phone System - General' started by Jonners59, Nov 23, 2013.

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  1. Jonners59

    Jonners59 New Member

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

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

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



    Code:
    23-Nov-2013 11:07:56.625   L:47.1[Extn] got Terminated Recv Req CANCEL from 192.168.1.90:5060 tid=798c9140c2c196ab1ca32280f60eff11 Call-ID=704268022@192_168_1_90:
    CANCEL sip:402086689100@192.168.1.51;user=phone SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK798c9140c2c196ab1ca32280f60eff11;rport=5060
    Max-Forwards: 70
    Contact: <sip:200@192.168.1.90:5060>
    To: <sip:402086689100@192.168.1.51;user=phone>
    From: "Baroni Limited"<sip:200@192.168.1.51>;tag=2294670366
    Call-ID: 704268022@192_168_1_90
    CSeq: 3 CANCEL
    Proxy-Authorization: Digest username="200", realm="3CXPhoneSystem", algorithm=MD5, uri="sip:402086689100@192.168.1.51;user=phone", nonce="414d535c08a11d6717:9207c2074ab942b3fd076ac4f35e9abc", response="cf81298bb9a8ef4fdd170651f4bdd233"
    User-Agent: N300 IP/42.075.00.000.000
    Content-Length: 0
    23-Nov-2013 11:07:56.625   Terminated from "Baroni Limited"<sip:200@192.168.1.51>;tag=2294670366 to <sip:402086689100@192.168.1.51;user=phone>;tag=fe23ec71; reason: RemoteBye
    23-Nov-2013 11:07:56.625   L:47.1[Extn] Sending: OnSendResp Send 487/INVITE from 0.0.0.0:0 tid=798c9140c2c196ab1ca32280f60eff11 Call-ID=704268022@192_168_1_90:
    SIP/2.0 487 Request Terminated
    Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK798c9140c2c196ab1ca32280f60eff11;rport=5060
    To: <sip:402086689100@192.168.1.51;user=phone>;tag=fe23ec71
    From: "Baroni Limited"<sip:200@192.168.1.51>;tag=2294670366
    Call-ID: 704268022@192_168_1_90
    CSeq: 3 INVITE
    Content-Length: 0
    23-Nov-2013 11:07:56.625   SendMsg from <sip:402086689100@192.168.1.51;user=phone>;tag=fe23ec71 to "Baroni Limited"<sip:200@192.168.1.51>;tag=2294670366
    23-Nov-2013 11:07:56.625   L:47.1[Extn] Sending: OnSendResp Send 200/CANCEL from 0.0.0.0:0 tid=798c9140c2c196ab1ca32280f60eff11 Call-ID=704268022@192_168_1_90:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK798c9140c2c196ab1ca32280f60eff11;rport=5060
    Contact: <sip:402086689100@192.168.1.51;user=phone>
    To: <sip:402086689100@192.168.1.51;user=phone>;tag=fe23ec71
    From: "Baroni Limited"<sip:200@192.168.1.51>;tag=2294670366
    Call-ID: 704268022@192_168_1_90
    CSeq: 3 CANCEL
    Content-Length: 0
    23-Nov-2013 11:07:56.625   SendMsg from <sip:402086689100@192.168.1.51;user=phone>;tag=fe23ec71 to "Baroni Limited"<sip:200@192.168.1.51>;tag=2294670366
    23-Nov-2013 11:07:52.345   L:47.2[Line:10001>>02086689100]: Terminating targets, reason: SIP ;cause=408 ;text="Request Timeout"
    23-Nov-2013 11:07:52.345   Leg L:47.2[Line:10001>>02086689100] is terminated: Cause: 408 Request Timeout/INVITE from local
    23-Nov-2013 11:07:52.345   L:47.2[Line:10001>>02086689100] got Terminated Recv 408/INVITE from 0.0.0.0:0 tid=470ece4f0b13026f Call-ID=MDIyN2RjZjc5MTRkNmNjZDM5MTQ5YjZlMjg1MTlhZTQ.:
    SIP/2.0 408 Request Timeout
    Via: SIP/2.0/UDP 192.168.1.51:5060;branch=z9hG4bK-d8754z-470ece4f0b13026f-1---d8754z-;rport
    To: <sip:02086689100@192.168.1.95:5062>;tag=17486b5c
    From: "Baroni Limited"<sip:6457@192.168.1.51:5060>;tag=dd4c2032
    Call-ID: MDIyN2RjZjc5MTRkNmNjZDM5MTQ5YjZlMjg1MTlhZTQ.
    CSeq: 1 INVITE
    Content-Length: 0
    23-Nov-2013 11:07:52.345   Terminated from <sip:02086689100@192.168.1.95:5062>;tag=17486b5c to "Baroni Limited"<sip:6457@192.168.1.51:5060>;tag=dd4c2032; reason: Error
    23-Nov-2013 11:07:52.345   ~Target=PSTNline:02086689100@(Ln.10001@OFFICE)
    23-Nov-2013 11:07:52.345   [CM503025]: Call(C:47): Calling T:Line:10005>>02086689100@[Dev:sip:844231085@voiptalk.org:5060] for L:47.1[Extn]
    23-Nov-2013 11:07:52.344   Route to L:47.3[Line:10005>>02086689100] sends Invite-OUT Send Req INVITE from 0.0.0.0:0 tid=c96f1d2adc69cd6a Call-ID=ZTkyMTU5MmQ4NjJlZjQ4NDZjMWNjOTIxMTliYjdlYmI.:
    INVITE sip:02086689100@voiptalk.org:5060 SIP/2.0
    Via: SIP/2.0/ ;branch=z9hG4bK-d8754z-c96f1d2adc69cd6a-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:844231085@82.14.176.51:5060>
    To: <sip:02086689100@voiptalk.org:5060>
    From: "Baroni Limited"<sip:844231085@voiptalk.org:5060>;tag=ed69824b
    Call-ID: ZTkyMTU5MmQ4NjJlZjQ4NDZjMWNjOTIxMTliYjdlYmI.
    CSeq: 1 INVITE
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
    Content-Type: application/sdp
    Supported: replaces
    Content-Length: 279
    
    v=0
    o=3cxPS 5351931904 123346092033 IN IP4 82.14.176.51
    s=3cxPS Audio call
    c=IN IP4 82.14.176.51
    t=0 0
    m=audio 9044 RTP/AVP 0 8 3 99 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:99 SPEEX/8000
    a=rtpmap:101 telephone-event/8000
    a=sendrecv
    23-Nov-2013 11:07:52.344   Dev(1):[sip:844231085@voiptalk.org:5060 / 844231085]: PBX contact is resolved to public IP: <sip:844231085@82.14.176.51:5060>
    23-Nov-2013 11:07:52.171   Outbound URI is used: sip:77.240.48.201:5065
    23-Nov-2013 11:07:52.167   Added leg L:47.3[Line:10005>>02086689100]
    23-Nov-2013 11:07:52.167   ~Route=Dev:sip:10001@192.168.1.95:5060
    23-Nov-2013 11:07:52.167   Call to T:Line:10001>>02086689100@[Dev:sip:10001@192.168.1.95:5060] from L:47.1[Extn] failed, cause: Cause: 408 Request Timeout/INVITE from local
    23-Nov-2013 11:07:52.163   L:47.2[Line:10001>>02086689100] got Failure: Failure Recv 408/INVITE from 0.0.0.0:0 tid=470ece4f0b13026f Call-ID=MDIyN2RjZjc5MTRkNmNjZDM5MTQ5YjZlMjg1MTlhZTQ.:
    SIP/2.0 408 Request Timeout
    Via: SIP/2.0/UDP 192.168.1.51:5060;branch=z9hG4bK-d8754z-470ece4f0b13026f-1---d8754z-;rport
    To: <sip:02086689100@192.168.1.95:5062>;tag=17486b5c
    From: "Baroni Limited"<sip:6457@192.168.1.51:5060>;tag=dd4c2032
    Call-ID: MDIyN2RjZjc5MTRkNmNjZDM5MTQ5YjZlMjg1MTlhZTQ.
    CSeq: 1 INVITE
    Content-Length: 0
    23-Nov-2013 11:07:52.157   [CM503003]: Call(C:47): Call to <sip:02086689100@192.168.1.95:5062> has failed; Cause: 408 Request Timeout/INVITE from local
    23-Nov-2013 11:07:52.155   Session 95886 has failed in leg L:47.2[Line:10001>>02086689100] ; Cause: 408 Request Timeout/INVITE from local
    23-Nov-2013 11:07:52.150   Failure from <sip:02086689100@192.168.1.95:5062>;tag=17486b5c to "Baroni Limited"<sip:6457@192.168.1.51:5060>;tag=dd4c2032
    23-Nov-2013 11:07:40.886   Currently active calls - 1: [47]
    23-Nov-2013 11:07:20.022   [CM503025]: Call(C:47): Calling T:Line:10001>>02086689100@[Dev:sip:10001@192.168.1.95:5060] for L:47.1[Extn]
    23-Nov-2013 11:07:20.022   Route to L:47.2[Line:10001>>02086689100] sends Invite-OUT Send Req INVITE from 0.0.0.0:0 tid=7609461399746f36 Call-ID=MDIyN2RjZjc5MTRkNmNjZDM5MTQ5YjZlMjg1MTlhZTQ.:
    INVITE sip:02086689100@192.168.1.95:5062 SIP/2.0
    Via: SIP/2.0/ ;branch=z9hG4bK-d8754z-7609461399746f36-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:6457@192.168.1.51:5060>
    To: <sip:02086689100@192.168.1.95:5062>
    From: "Baroni Limited"<sip:6457@192.168.1.51:5060>;tag=dd4c2032
    Call-ID: MDIyN2RjZjc5MTRkNmNjZDM5MTQ5YjZlMjg1MTlhZTQ.
    CSeq: 1 INVITE
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
    Content-Type: application/sdp
    Supported: replaces
    Content-Length: 357
    
    v=0
    o=3cxPS 441525993472 356364845057 IN IP4 192.168.1.51
    s=3cxPS Audio call
    c=IN IP4 192.168.1.51
    t=0 0
    m=audio 7170 RTP/AVP 0 8 3 13 9 110 99 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:13 CN/8000
    a=rtpmap:9 G722/8000
    a=rtpmap:110 iLBC/8000
    a=rtpmap:99 SPEEX/8000
    a=rtpmap:101 telephone-event/8000
    a=sendrecv
    23-Nov-2013 11:07:19.996   Outbound URI is used: sip:10001@192.168.1.95:5060
    23-Nov-2013 11:07:19.983   Added leg L:47.2[Line:10001>>02086689100]
    23-Nov-2013 11:07:19.927   [Flow] Call(C:47): making call from L:47.1[Extn] to T:Line:10002>>02086689100@[Dev:sip:10002@127.0.0.1:5060]
    23-Nov-2013 11:07:19.927   [CM503027]: Call(C:47): From: Extn:200 ("Baroni Limited" <sip:200@192.168.1.51:5060>)  to  T:Line:10002>>02086689100@[Dev:sip:10002@127.0.0.1:5060]
    23-Nov-2013 11:07:19.927   [CM503004]: Call(C:47): Route 3: from L:47.1[Extn] to T:Line:10002>>02086689100@[Dev:sip:10002@127.0.0.1:5060]
    23-Nov-2013 11:07:19.927   Line limit check: Current # of calls for line Lc:10002(@HOME[<sip:10002@127.0.0.1:5060>]) is 0; limit is 1
    23-Nov-2013 11:07:19.927   [Flow] Call(C:47): making call from L:47.1[Extn] to T:Line:10005>>02086689100@[Dev:sip:844231085@voiptalk.org:5060]
    23-Nov-2013 11:07:19.927   [CM503027]: Call(C:47): From: Extn:200 ("Baroni Limited" <sip:200@192.168.1.51:5060>)  to  T:Line:10005>>02086689100@[Dev:sip:844231085@voiptalk.org:5060]
    23-Nov-2013 11:07:19.927   [CM503004]: Call(C:47): Route 2: from L:47.1[Extn] to T:Line:10005>>02086689100@[Dev:sip:844231085@voiptalk.org:5060]
    23-Nov-2013 11:07:19.927   Line limit check: Current # of calls for line Lc:10005(@VoIPTalk[<sip:844231085@voiptalk.org:5060>]) is 0; limit is 2
    23-Nov-2013 11:07:19.927   [Flow] Call(C:47): making call from L:47.1[Extn] to T:Line:10001>>02086689100@[Dev:sip:10001@192.168.1.95:5060]
    23-Nov-2013 11:07:19.927   [CM503027]: Call(C:47): From: Extn:200 ("Baroni Limited" <sip:200@192.168.1.51:5060>)  to  T:Line:10001>>02086689100@[Dev:sip:10001@192.168.1.95:5060]
    23-Nov-2013 11:07:19.927   [CM503004]: Call(C:47): Route 1: from L:47.1[Extn] to T:Line:10001>>02086689100@[Dev:sip:10001@192.168.1.95:5060]
    23-Nov-2013 11:07:19.927   Line limit check: Current # of calls for line Lc:10001(@OFFICE[<sip:10001@192.168.1.95:5060>]) is 0; limit is 1
    23-Nov-2013 11:07:19.927   Call(C:47): Call from Extn:200 to 402086689100 matches outbound rule 'PSTN Baroni'
    23-Nov-2013 11:07:19.927   [Flow] Call(C:47): has built target endpoint: Out#:>>Rule{PSTN Baroni}>>402086689100 for call from L:47.1[Extn]
    23-Nov-2013 11:07:19.927   [Flow] Target endpoint for 402086689100 is Out#:>>Rule{PSTN Baroni}>>402086689100
    23-Nov-2013 11:07:19.927   Selected prefix: 4
    23-Nov-2013 11:07:19.927   Looking for outbound rule: dialed = [402086689100], processed: [402086689100]; from-ext:
    23-Nov-2013 11:07:19.927   [Flow] Building target endpoint to 402086689100 from "Baroni Limited" <sip:200@192.168.1.51:5060>
    23-Nov-2013 11:07:19.927   [CM503010]: Call(C:47): Making route(s) from Extn:200 to <sip:402086689100@192.168.1.51:5060>
    23-Nov-2013 11:07:19.927   Remote SDP is set for leg L:47.1[Extn]
    23-Nov-2013 11:07:19.927   OnOffer from "Baroni Limited"<sip:200@192.168.1.51>;tag=2294670366
    23-Nov-2013 11:07:19.926   [CM505001]: Endpoint Extn:200: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [N300 IP/42.075.00.000.000] PBX contact: [sip:200@192.168.1.51:5060]
    23-Nov-2013 11:07:19.905   Inbound DID: ''; Phonebook Name: ''
    23-Nov-2013 11:07:19.905   [CM500002]: Call(C:47): Info on incoming INVITE from Extn:200:
    Invite-IN Recv Req INVITE from 192.168.1.90:5060 tid=798c9140c2c196ab1ca32280f60eff11 Call-ID=704268022@192_168_1_90:
    INVITE sip:402086689100@192.168.1.51;user=phone SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK798c9140c2c196ab1ca32280f60eff11;rport=5060
    Max-Forwards: 70
    Contact: <sip:200@192.168.1.90:5060>
    To: <sip:402086689100@192.168.1.51;user=phone>
    From: "Baroni Limited"<sip:200@192.168.1.51>;tag=2294670366
    Call-ID: 704268022@192_168_1_90
    CSeq: 3 INVITE
    Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
    Content-Type: application/sdp
    Proxy-Authorization: Digest username="200",realm="3CXPhoneSystem",algorithm=MD5,uri="sip:402086689100@192.168.1.51;user=phone",nonce="414d535c08a11d6717:9207c2074ab942b3fd076ac4f35e9abc",response="371787970a59a02275636051f2acc0d6"
    Supported: replaces
    User-Agent: N300 IP/42.075.00.000.000
    Allow-Events: message-summary, refer, ua-profile, talk
    Content-Length: 381
    
    v=0
    o=200 5006 197 IN IP4 192.168.1.90
    s=Mapping
    c=IN IP4 192.168.1.90
    t=0 0
    m=audio 5006 RTP/AVP 9 8 0 96 97 2 18 101
    a=rtpmap:9 G722/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:96 G726-32/8000
    a=rtpmap:97 AAL2-G726-32/8000
    a=rtpmap:2 G726-32/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    23-Nov-2013 11:07:19.905   [CM503001]: Call(C:47): Incoming call from Extn:200 to <sip:402086689100@192.168.1.51:5060>
    23-Nov-2013 11:07:19.868   Outbound URI is used: sip:200@192.168.1.90:5060
    23-Nov-2013 11:07:19.868   IncomingCall: C:47 from <sip:200@192.168.1.51:5060> to <sip:402086689100@192.168.1.51:5060>
    23-Nov-2013 11:07:19.868   Added leg L:C:47.1[No endpoint yet]
    23-Nov-2013 11:07:19.865   UasSession 95857 started
    23-Nov-2013 11:07:19.865   Call from "Baroni Limited"<sip:200@192.168.1.51>;tag=2294670366 to <sip:402086689100@192.168.1.51;user=phone>;tag=fe23ec71
    23-Nov-2013 11:07:10.769   Currently active calls [none]
     
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  2. bardissi

    bardissi Member

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    I don't have any spa3102 but did hook up a PAP2T (pre spa3102) and that works fine manually provisioning this with the extension info.
     
  3. leejor

    leejor Well-Known Member

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    There are a number of settings that need to change from the defaults if you want to use the FXS port as an extension AND the FXO port at a 3CX trunk AND keep the two separated. It's all well and good to plug in a phone and dial out directly on the FXO port, but then why have 3CX, right?

    Be sure that the FXO port on the 3102 has 5062 as the assigned port. That's the one 3CX want to use even though the 3102 default is 5061.

    If you continue to have issues, post the PSTN tab (to start with) and I'll look over your settings.
     
  4. Jonners59

    Jonners59 New Member

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    Thanks guys
    I have used the 3CX, with Grandstream for many years without trouble, so was not expecting any difficulties, and due to the time since I last had to do configs I am a bit rusty, besides I am not a techie....

    I use the 3CX because I work from home for myslef I have a home line and work line that I want to keep seperate but also give a professional feel to the company call handling.

    The 2 x Cisco devices are to replace the faulty Grandstream. Should have been able to just plug in relly but nothing seems to be working the same: The devices are to be used, not as an FXS or ATA, but as a gateway FXO... the idea being calls on my home line go to 10001 (6457) and calls to my office line go to 10002 (9100)... and each has its own ring groups and IVR settings.

    6457 and 9100 being the lat 4 digits of my PSTN lines, this is new to me as it was not on v9 of 3CX

    Just because my wife uses a code for cheaper calls that are associated to the home line and for local services I have set up outbound codes for routing out over the FXO to PSTN from the 3CX and on failover calls can go out via the FXO to the PSTN (as I use a Siemens IP DECT phone).

    OK, less of the chat.

    I have already done the 5062 port bit.... so not that.

    I have therefore attached screen shots. I have done more than asked for just in case, better more then less. I have attached the SIP Tab, Regional Tab, Line Tab and PSTN Tabs... As my screen will not hold the whole page I have had to do multiple per tab, 3 for the all imprtant PSTN, sorry. I hope that gives you some cluse.

    And thank you for your help

    Attached are screen shots
     

    Attached Files:

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  5. leejor

    leejor Well-Known Member

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    If you don't plan on using the ATA portion , then disable it (which you have done). In fact there is no reason to plug a phone into there at all, it just confuses things.

    Dialplan 8 should not have the IP attached at the end. The 3102 already knows where to send the calls.

    You also didn't need to datafill the gateway accounts info. Or, the dialplan just after that. That dialplan can be used for a set (FXS port) that is an extension off 3CX.

    You don't need to put in an outbound proxy, but is shouldn't hurt.

    Get rid of the * in the caller ID pattern, that has been known to cause issues.

    Delete the VoIP Users and Passwords (HTTP Authentication), not necessary.

    You also don't need to enable STUN as everything is on the same LAN.
     
  6. Jonners59

    Jonners59 New Member

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    Thanks Leejor
    Made the changes, but sadly still does not work.

    On inbound calls the gateway says "VOIP Gateway Call" when a call is in progress but it does not register on the 3CX......... see attached.

    Outbound does nothing at all.

    Any other ideas, please?

    Code:
    23-Nov-2013 23:39:59.815   Currently active calls [none]
    23-Nov-2013 23:39:36.770   Currently active calls [none]
    23-Nov-2013 23:39:21.374   L:12.2[Line:10005>>02086606457]: Terminating targets, reason: SIP ;cause=487 ;text="Request Terminated"
    23-Nov-2013 23:39:21.374   Leg L:12.2[Line:10005>>02086606457] is terminated: Cause: 487 Request Terminated/INVITE from 77.240.48.201:5065
    23-Nov-2013 23:39:21.374   L:12.2[Line:10005>>02086606457] got Terminated Recv 487/INVITE from 77.240.48.201:5065 tid=391a7e750e1bc63c Call-ID=NmY0YTNhZGJlMDUyMmU2Mzk5MGE5YmI3M2NhNWZiNmQ.:
    SIP/2.0 487 Request Terminated
    Via: SIP/2.0/UDP 192.168.1.51:5060;received=82.14.176.51;branch=z9hG4bK-d8754z-391a7e750e1bc63c-1---d8754z-;rport=5060
    To: <sip:02086606457@voiptalk.org:5060>;tag=as7497ea4b
    From: "Baroni Limited"<sip:844231085@voiptalk.org:5060>;tag=fc2f3667
    Call-ID: NmY0YTNhZGJlMDUyMmU2Mzk5MGE5YmI3M2NhNWZiNmQ.
    CSeq: 2 INVITE
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Server: voip
    Supported: replaces
    Content-Length: 0
    23-Nov-2013 23:39:21.374   Terminated from <sip:02086606457@voiptalk.org:5060>;tag=as7497ea4b to "Baroni Limited"<sip:844231085@voiptalk.org:5060>;tag=fc2f3667; reason: LocalCancel
    23-Nov-2013 23:39:21.328   [CM503008]: Call(C:12): Call is terminated
    23-Nov-2013 23:39:21.328   L:12.2[Line:10005>>02086606457]: Terminating targets, reason: SIP ;cause=200 ;text="Call terminated on user request"
    23-Nov-2013 23:39:21.328   Blocking refers for the Call(C:12)
    23-Nov-2013 23:39:21.328   Call(C:12) is terminated
    23-Nov-2013 23:39:21.325   ~Target=PSTNline:02086606457@(Ln.10002@HOME)
    23-Nov-2013 23:39:21.325   ~Route=Dev:sip:10002@192.168.100.95:5062
    23-Nov-2013 23:39:21.325   ~Target=PSTNline:02086606457@(Ln.10001@OFFICE)
    23-Nov-2013 23:39:21.325   ~Route=Dev:sip:10001@127.0.0.1:5062
    23-Nov-2013 23:39:21.325   ~Target=VoIPline:02086606457@(Ln.10005@VoIPTalk)
    23-Nov-2013 23:39:21.325   ~Route=Dev:sip:844231085@voiptalk.org:5060
    23-Nov-2013 23:39:21.325   L:12.2[Line:10005>>02086606457]: Terminating targets, reason: SIP ;cause=487 ;text="Request Terminated"
    23-Nov-2013 23:39:21.317   Reason: SIP ;cause=487 ;text="Request Terminated"
    23-Nov-2013 23:39:21.317   L:12.1[Extn]: Terminating targets, reason: SIP ;cause=487 ;text="Request Terminated"
    23-Nov-2013 23:39:21.317   Leg L:12.1[Extn] is terminated: Cause: CANCEL from 192.168.1.90:5060
    23-Nov-2013 23:39:21.308   L:12.1[Extn] got Terminated Recv Req CANCEL from 192.168.1.90:5060 tid=9835c633f7351bf6a170b30163752842 Call-ID=1156366866@192_168_1_90:
    CANCEL sip:02086606457@192.168.1.51;user=phone SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK9835c633f7351bf6a170b30163752842;rport=5060
    Max-Forwards: 70
    Contact: <sip:200@192.168.1.90:5060>
    To: <sip:02086606457@192.168.1.51;user=phone>
    From: "Baroni Limited"<sip:200@192.168.1.51>;tag=2677598368
    Call-ID: 1156366866@192_168_1_90
    CSeq: 3 CANCEL
    Proxy-Authorization: Digest username="200", realm="3CXPhoneSystem", algorithm=MD5, uri="sip:02086606457@192.168.1.51;user=phone", nonce="414d535c08a1cd8d17:717b60782fd39a3ecdc934fc4fc1481a", response="07b68a8edd56ea95b4d563d2f6cf4255"
    User-Agent: N300 IP/42.075.00.000.000
    Content-Length: 0
    23-Nov-2013 23:39:21.308   Terminated from "Baroni Limited"<sip:200@192.168.1.51>;tag=2677598368 to <sip:02086606457@192.168.1.51;user=phone>;tag=3a14d305; reason: RemoteBye
    23-Nov-2013 23:39:21.308   L:12.1[Extn] Sending: OnSendResp Send 487/INVITE from 0.0.0.0:0 tid=9835c633f7351bf6a170b30163752842 Call-ID=1156366866@192_168_1_90:
    SIP/2.0 487 Request Terminated
    Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK9835c633f7351bf6a170b30163752842;rport=5060
    To: <sip:02086606457@192.168.1.51;user=phone>;tag=3a14d305
    From: "Baroni Limited"<sip:200@192.168.1.51>;tag=2677598368
    Call-ID: 1156366866@192_168_1_90
    CSeq: 3 INVITE
    Content-Length: 0
    23-Nov-2013 23:39:21.308   SendMsg from <sip:02086606457@192.168.1.51;user=phone>;tag=3a14d305 to "Baroni Limited"<sip:200@192.168.1.51>;tag=2677598368
    23-Nov-2013 23:39:21.308   L:12.1[Extn] Sending: OnSendResp Send 200/CANCEL from 0.0.0.0:0 tid=9835c633f7351bf6a170b30163752842 Call-ID=1156366866@192_168_1_90:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK9835c633f7351bf6a170b30163752842;rport=5060
    Contact: <sip:02086606457@192.168.1.51;user=phone>
    To: <sip:02086606457@192.168.1.51;user=phone>;tag=3a14d305
    From: "Baroni Limited"<sip:200@192.168.1.51>;tag=2677598368
    Call-ID: 1156366866@192_168_1_90
    CSeq: 3 CANCEL
    Content-Length: 0
    23-Nov-2013 23:39:21.308   SendMsg from <sip:02086606457@192.168.1.51;user=phone>;tag=3a14d305 to "Baroni Limited"<sip:200@192.168.1.51>;tag=2677598368
    23-Nov-2013 23:39:19.453   Currently active calls - 1: [12]
    23-Nov-2013 23:39:04.416   Currently active calls - 1: [12]
    23-Nov-2013 23:38:57.058   L:12.2[Line:10005>>02086606457] got EarlyMedia Recv 183/INVITE from 77.240.48.201:5065 tid=391a7e750e1bc63c Call-ID=NmY0YTNhZGJlMDUyMmU2Mzk5MGE5YmI3M2NhNWZiNmQ.:
    SIP/2.0 183 Session Progress
    Via: SIP/2.0/UDP 192.168.1.51:5060;received=82.14.176.51;branch=z9hG4bK-d8754z-391a7e750e1bc63c-1---d8754z-;rport=5060
    Record-Route: <sip:77.240.48.94;lr=on;ftag=fc2f3667>
    Record-Route: <sip:xuser@77.240.48.201;r2=on;lr=on;ftag=fc2f3667>
    Record-Route: <sip:xuser@77.240.48.201:5065;r2=on;lr=on;ftag=fc2f3667>
    Contact: <sip:CALL-74336640-02086606457@77.240.54.23:5061>
    To: <sip:02086606457@voiptalk.org:5060>;tag=as7497ea4b
    From: "Baroni Limited"<sip:844231085@voiptalk.org:5060>;tag=fc2f3667
    Call-ID: NmY0YTNhZGJlMDUyMmU2Mzk5MGE5YmI3M2NhNWZiNmQ.
    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 93155214 93155214 IN IP4 77.240.54.23
    s=voip
    c=IN IP4 77.240.48.209
    t=0 0
    m=audio 41796 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
    23-Nov-2013 23:38:57.057   EarlyMedia from <sip:02086606457@voiptalk.org:5060>;tag=as7497ea4b to "Baroni Limited"<sip:844231085@voiptalk.org:5060>;tag=fc2f3667
    23-Nov-2013 23:38:57.057   L:12.1[Extn] Sending: OnSendResp Send 180/INVITE from 0.0.0.0:0 tid=9835c633f7351bf6a170b30163752842 Call-ID=1156366866@192_168_1_90:
    SIP/2.0 180 Ringing
    Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK9835c633f7351bf6a170b30163752842;rport=5060
    Contact: <sip:02086606457@192.168.1.51;user=phone>
    To: <sip:02086606457@192.168.1.51;user=phone>;tag=3a14d305
    From: "Baroni Limited"<sip:200@192.168.1.51>;tag=2677598368
    Call-ID: 1156366866@192_168_1_90
    CSeq: 3 INVITE
    Content-Length: 0
    23-Nov-2013 23:38:57.055   SendMsg from <sip:02086606457@192.168.1.51;user=phone>;tag=3a14d305 to "Baroni Limited"<sip:200@192.168.1.51>;tag=2677598368
    23-Nov-2013 23:38:57.051   Provisional response arrived for session 5614 of Leg L:12.2[Line:10005>>02086606457]
    23-Nov-2013 23:38:57.051   L:12.2[Line:10005>>02086606457] got Provisional Recv 183/INVITE from 77.240.48.201:5065 tid=391a7e750e1bc63c Call-ID=NmY0YTNhZGJlMDUyMmU2Mzk5MGE5YmI3M2NhNWZiNmQ.:
    SIP/2.0 183 Session Progress
    Via: SIP/2.0/UDP 192.168.1.51:5060;received=82.14.176.51;branch=z9hG4bK-d8754z-391a7e750e1bc63c-1---d8754z-;rport=5060
    Record-Route: <sip:77.240.48.94;lr=on;ftag=fc2f3667>
    Record-Route: <sip:xuser@77.240.48.201;r2=on;lr=on;ftag=fc2f3667>
    Record-Route: <sip:xuser@77.240.48.201:5065;r2=on;lr=on;ftag=fc2f3667>
    Contact: <sip:CALL-74336640-02086606457@77.240.54.23:5061>
    To: <sip:02086606457@voiptalk.org:5060>;tag=as7497ea4b
    From: "Baroni Limited"<sip:844231085@voiptalk.org:5060>;tag=fc2f3667
    Call-ID: NmY0YTNhZGJlMDUyMmU2Mzk5MGE5YmI3M2NhNWZiNmQ.
    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 93155214 93155214 IN IP4 77.240.54.23
    s=voip
    c=IN IP4 77.240.48.209
    t=0 0
    m=audio 41796 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
    23-Nov-2013 23:38:57.051   Provisional(183) from <sip:02086606457@voiptalk.org:5060>;tag=as7497ea4b to "Baroni Limited"<sip:844231085@voiptalk.org:5060>;tag=fc2f3667
    23-Nov-2013 23:38:57.050   [CM505003]: Provider:[VoIPTalk] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] PBX contact: [sip:844231085@82.14.176.51:5060]
    23-Nov-2013 23:38:57.050   [CM503002]: Call(C:12): Alerting Line:10005>>02086606457 by contact <sip:844231085@voiptalk.org:5060>
    23-Nov-2013 23:38:57.034   Phonebook entry not found for number 02086606457
    23-Nov-2013 23:38:57.034   Looking for phone number 02086606457 in tenant's 'default' phonebook
    23-Nov-2013 23:38:57.023   UacSession 5614 has formed leg L:12.2[Line:10005>>02086606457]
    23-Nov-2013 23:38:57.023   Answer from <sip:02086606457@voiptalk.org:5060>;tag=as7497ea4b to "Baroni Limited"<sip:844231085@voiptalk.org:5060>;tag=fc2f3667
    23-Nov-2013 23:38:54.487   [CM503025]: Call(C:12): Calling T:Line:10005>>02086606457@[Dev:sip:844231085@voiptalk.org:5060] for L:12.1[Extn]
    23-Nov-2013 23:38:54.487   Route to L:12.2[Line:10005>>02086606457] sends Invite-OUT Send Req INVITE from 0.0.0.0:0 tid=4768cd30af792c16 Call-ID=NmY0YTNhZGJlMDUyMmU2Mzk5MGE5YmI3M2NhNWZiNmQ.:
    INVITE sip:02086606457@voiptalk.org:5060 SIP/2.0
    Via: SIP/2.0/ ;branch=z9hG4bK-d8754z-4768cd30af792c16-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:844231085@82.14.176.51:5060>
    To: <sip:02086606457@voiptalk.org:5060>
    From: "Baroni Limited"<sip:844231085@voiptalk.org:5060>;tag=fc2f3667
    Call-ID: NmY0YTNhZGJlMDUyMmU2Mzk5MGE5YmI3M2NhNWZiNmQ.
    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 211359367168 238270021633 IN IP4 82.14.176.51
    s=3cxPS Audio call
    c=IN IP4 82.14.176.51
    t=0 0
    m=audio 9020 RTP/AVP 0 8 3 99 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:99 SPEEX/8000
    a=rtpmap:101 telephone-event/8000
    a=sendrecv
    23-Nov-2013 23:38:54.481   Dev(1):[sip:844231085@voiptalk.org:5060 / 844231085]: PBX contact is resolved to public IP: <sip:844231085@82.14.176.51:5060>
    23-Nov-2013 23:38:54.300   Outbound URI is used: sip:77.240.48.201:5065
    23-Nov-2013 23:38:54.295   Added leg L:12.2[Line:10005>>02086606457]
    23-Nov-2013 23:38:54.280   [Flow] Call(C:12): making call from L:12.1[Extn] to T:Line:10002>>02086606457@[Dev:sip:10002@192.168.100.95:5062]
    23-Nov-2013 23:38:54.280   [CM503027]: Call(C:12): From: Extn:200 ("Baroni Limited" <sip:200@192.168.1.51:5060>)  to  T:Line:10002>>02086606457@[Dev:sip:10002@192.168.100.95:5062]
    23-Nov-2013 23:38:54.280   [CM503004]: Call(C:12): Route 3: from L:12.1[Extn] to T:Line:10002>>02086606457@[Dev:sip:10002@192.168.100.95:5062]
    23-Nov-2013 23:38:54.280   Line limit check: Current # of calls for line Lc:10002(@HOME[<sip:10002@192.168.100.95:5062>]) is 0; limit is 1
    23-Nov-2013 23:38:54.280   [Flow] Call(C:12): making call from L:12.1[Extn] to T:Line:10001>>02086606457@[Dev:sip:10001@127.0.0.1:5062]
    23-Nov-2013 23:38:54.280   [CM503027]: Call(C:12): From: Extn:200 ("Baroni Limited" <sip:200@192.168.1.51:5060>)  to  T:Line:10001>>02086606457@[Dev:sip:10001@127.0.0.1:5062]
    23-Nov-2013 23:38:54.280   [CM503004]: Call(C:12): Route 2: from L:12.1[Extn] to T:Line:10001>>02086606457@[Dev:sip:10001@127.0.0.1:5062]
    23-Nov-2013 23:38:54.280   Line limit check: Current # of calls for line Lc:10001(@OFFICE[<sip:10001@127.0.0.1:5062>]) is 0; limit is 1
    23-Nov-2013 23:38:54.280   [Flow] Call(C:12): making call from L:12.1[Extn] to T:Line:10005>>02086606457@[Dev:sip:844231085@voiptalk.org:5060]
    23-Nov-2013 23:38:54.280   [CM503027]: Call(C:12): From: Extn:200 ("Baroni Limited" <sip:200@192.168.1.51:5060>)  to  T:Line:10005>>02086606457@[Dev:sip:844231085@voiptalk.org:5060]
    23-Nov-2013 23:38:54.280   [CM503004]: Call(C:12): Route 1: from L:12.1[Extn] to T:Line:10005>>02086606457@[Dev:sip:844231085@voiptalk.org:5060]
    23-Nov-2013 23:38:54.262   Line limit check: Current # of calls for line Lc:10005(@VoIPTalk[<sip:844231085@voiptalk.org:5060>]) is 0; limit is 2
    23-Nov-2013 23:38:54.250   Call(C:12): Call from Extn:200 to 02086606457 matches outbound rule 'Outbound Baroni'
    23-Nov-2013 23:38:54.250   [Flow] Call(C:12): has built target endpoint: Out#:>>Rule{Outbound Baroni}>>02086606457 for call from L:12.1[Extn]
    23-Nov-2013 23:38:54.250   [Flow] Target endpoint for 02086606457 is Out#:>>Rule{Outbound Baroni}>>02086606457
    23-Nov-2013 23:38:54.250   Caller '200' is found in the list of allowed extensions: 200-204
    23-Nov-2013 23:38:54.250   Selected prefix: 0
    23-Nov-2013 23:38:54.250   Looking for outbound rule: dialed = [02086606457], processed: [02086606457]; from-ext:
    23-Nov-2013 23:38:54.233   [Flow] Building target endpoint to 02086606457 from "Baroni Limited" <sip:200@192.168.1.51:5060>
    23-Nov-2013 23:38:54.233   [CM503010]: Call(C:12): Making route(s) from Extn:200 to <sip:02086606457@192.168.1.51:5060>
    23-Nov-2013 23:38:54.233   Remote SDP is set for leg L:12.1[Extn]
    23-Nov-2013 23:38:54.233   OnOffer from "Baroni Limited"<sip:200@192.168.1.51>;tag=2677598368
    23-Nov-2013 23:38:54.233   [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:200@192.168.1.51:5060]
    23-Nov-2013 23:38:54.216   Inbound DID: ''; Phonebook Name: ''
    23-Nov-2013 23:38:54.216   [CM500002]: Call(C:12): Info on incoming INVITE from Extn:200:
    Invite-IN Recv Req INVITE from 192.168.1.90:5060 tid=9835c633f7351bf6a170b30163752842 Call-ID=1156366866@192_168_1_90:
    INVITE sip:02086606457@192.168.1.51;user=phone SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK9835c633f7351bf6a170b30163752842;rport=5060
    Max-Forwards: 70
    Contact: <sip:200@192.168.1.90:5060>
    To: <sip:02086606457@192.168.1.51;user=phone>
    From: "Baroni Limited"<sip:200@192.168.1.51>;tag=2677598368
    Call-ID: 1156366866@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:02086606457@192.168.1.51;user=phone",nonce="414d535c08a1cd8d17:717b60782fd39a3ecdc934fc4fc1481a",response="d8635d98cf960f4278cc8356fd9c5666"
    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 5010 217 IN IP4 192.168.1.90
    s=Mapping
    c=IN IP4 192.168.1.90
    t=0 0
    m=audio 5010 RTP/AVP 9 8 0 96 97 2 18 101
    a=rtpmap:9 G722/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:96 G726-32/8000
    a=rtpmap:97 AAL2-G726-32/8000
    a=rtpmap:2 G726-32/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    23-Nov-2013 23:38:54.216   [CM503001]: Call(C:12): Incoming call from Extn:200 to <sip:02086606457@192.168.1.51:5060>
    23-Nov-2013 23:38:54.062   Outbound URI is used: sip:200@192.168.1.90:5060
    23-Nov-2013 23:38:54.058   IncomingCall: C:12 from <sip:200@192.168.1.51:5060> to <sip:02086606457@192.168.1.51:5060>
    23-Nov-2013 23:38:54.058   Added leg L:C:12.1[No endpoint yet]
    23-Nov-2013 23:38:54.055   UasSession 5611 started
    23-Nov-2013 23:38:54.054   Call from "Baroni Limited"<sip:200@192.168.1.51>;tag=2677598368 to <sip:02086606457@192.168.1.51;user=phone>;tag=3a14d305
    23-Nov-2013 23:38:49.379   Currently active calls [none]
    23-Nov-2013 23:38:34.362   Currently active calls [none]
    23-Nov-2013 23:38:24.245   Registered line: Lc:10001(@OFFICE[<sip:10001@127.0.0.1:5062>])
    23-Nov-2013 23:38:24.225   Dev(188):[sip:10001@127.0.0.1:5062 / 10001] has updated source address: 192.168.1.96:5062
    23-Nov-2013 23:38:24.223   Line 10001 got Device: Dev(187):[sip:10001@127.0.0.1:5062 / 10001]
    23-Nov-2013 23:38:24.214   Dev(187):[sip:10001@127.0.0.1:5062 / 10001] has updated source address: 192.168.1.96:5062
    23-Nov-2013 23:38:19.054   Currently active calls [none]
    23-Nov-2013 23:38:03.774   Currently active calls [none]
    23-Nov-2013 23:37:48.729   Currently active calls [none]
    23-Nov-2013 23:37:33.709   Currently active calls [none]
     

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  7. leejor

    leejor Well-Known Member

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    I'm assuming that you assigned a fixed IP to each of the 3102's?

    Did you put those IP's into the trunk settings in 3CX? The above logs should show the calls going to the Gateway IP:5062
     
  8. Jonners59

    Jonners59 New Member

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    Leejor
    Good morning and thank you. Yes to q 1: 192.168.1.95 & 96

    Q2: Is as per the attached to what you refer? If not where and how do I add them, please?

    Again to give a full picture I have added further screenshots of other settings on the 3CX re Gateway setup, guess the Gateway config means little without the settings on the 3CX itself :)

    Many thanks
     

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  9. Cjay

    Cjay New Member

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    I believe Leejor is in a time zone where he should be getting a bit of shut-eye right now, so as a quick answer to q2 to give you 24x7 support:
    Under the VoIP/PSTN Gateway section you will see entry for your 3102's where IP, port & trunk ID are input - see my attached sample. What do you have here?

    [By the way I have the gateway port set to 5061 since I am also using the FXS port on the 3102 to drive an analogue phone using port 5060]

    Regds,
    Chris
     

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  10. Jonners59

    Jonners59 New Member

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    Good morning Chris and thank you....

    Seems I was correcting and adding to my reply when you sent your message. As you can see from the above, the answer to Q2 is yes. I did for a moment wonder about the port, but that matches too, so it is not that. This is getting very frustrating as I can not see why it does not work, stating the obvious.

    I use a Draytech 2820 router and have the QoS and Port redirects set up, but that should not matter. What else could it be?
     
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  11. Cjay

    Cjay New Member

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    What do you see under the 3102 Voice/Info 'PSTN Line Status' section - Does it show anything ?

    Incidentally my DP8 is just (S0:10000), no reference to IP.
     
  12. Jonners59

    Jonners59 New Member

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    Charles screenshot of one attached
    Gateway seems to see the call comming in, but does not seem to be able to route it to the 3CX, and outbound the 3 CX does not seem to be able to send to the Gateway.

    I have also noted that the registration seems to sometimes "unregister" for a period. Don't ever rcall that happening before I upgraded to V12

    Yes, I tired with and without the IP. neither worked. I have been souring the web and betting all sorts of settings, tried everything in desperation!!!!
     

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  13. Jonners59

    Jonners59 New Member

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    Help, please, anyone....... :cry:
     
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  14. leejor

    leejor Well-Known Member

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    Normally, what I suggest is a factory reset, and a fresh start. Given that you are not located in North America where the default settings cater. We'll try to work with what we have.

    Have you read over this page? If you follow it through, you should have no problems. As I said before, just note 22e, regarding the use of the *.
    http: //www.3cx.com/voip-gateways/linksys-spa3102/ - NO LONGER AVAILABLE


    If you are starting from scratch, and follow this, it should work. Just be sure to change your regional settings to accommodate UK caller ID and other (unique) settings.

    For incoming calls to work, dialplan 8 must be set to (S0<:10001>) (or whatever trunk number you are using.) Note... that is a zero after the S.

    The 3102 has a lot of settings that many people will never make use of, and can be extremely frustrating as many have found. But, it does work, and does a very good job when set-up correctly.

    You don't really need to make use of DID settings for this gateway as it simply gets an incoming call, it isn't a VoIP trunk that also receives the called number, and routes on that. In a "basic" set-up any incoming calls can be sent to a single extension, or a ring group
     
  15. Jonners59

    Jonners59 New Member

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    Good morning Leejor
    Thanks for yesterday.

    I have done man resets, and makes no difference. Also tried the link you mentioned, even better for UK peeps is this one:
    http://www.aoakley.com/articles/2008-01-08.php... designed for noddies lik me.

    Yes, have also used a 0 (zero) and reverted back to taking out the IP address.

    Have just taken out the DID...

    I made a few test calls a moment ago, before I read your reply and removed the DID. Thought I would test under fall back at least be able to get some calls to and out the house. Interesting results!

    Without power no problem sending and receiving calls
    With power, but no Ethernet one of the spa can make a call (HOME), but goes to the operator and the other (OFFICE) dials out but does not ring at the other end - tried the HOME and my mobile....... Am going to check for differences and feedback, am also going to turn off the IP and MAC Bind, just in case.

    Thank you
    Jonners
     
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  16. complex1

    complex1 Active Member

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  17. Cjay

    Cjay New Member

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    I can send you my complete 3102 config as .htm if this would help (don't think I can 'attach' on this forum). I notice you didn't have some of the UK (BT?) specific items correct. PM with an email address if you want.
    Chris
     
  18. Jonners59

    Jonners59 New Member

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    Complex1 thank you... Your system is different to mine, so was tricky to check. Big difference was you put the port in the address, tried it but did not work, sadly, but thanks

    Chris
    Yes please, pinging now.....
    PS: what bits of BT am I missing.....?

    Jonners
     
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  19. Cjay

    Cjay New Member

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    PSTN disconnect detection and some of the timers (no need for answer delay with BT CLID as FSK is sent prior to first ring).

    Chris
     
  20. complex1

    complex1 Active Member

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    Jonners,

    Send you a PM.
     
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