Setting up Linksys SPA3102 for FAX

Discussion in '3CX Phone System - General' started by vgermscheid, Feb 7, 2012.

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  1. vgermscheid

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    I've read through a number of articles, including http: //www.3cx.com/voip-gateways/linksys-3102/ - NO LONGER AVAILABLE but they don't seem to quite get me there. The device seems to be set up correctly, I can dial to/from it internally and can dial into it from an external line, but when I try dialing an external number, I just get a busy signal.

    Any suggestions?
     
  2. leejor

    leejor Well-Known Member

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    It could be settings in the 3102, or it could be your outbound rules, or it could be the ATA/set doing the dialling.
    Post, or look at, the 3CX log, showing a failed call from start to finish.
    Are you dialling from a 3CX extension or from the FXS port (directly out on the FXO port), on the 3102?
     
  3. vgermscheid

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    Here are the log messages.

    In the event log:
    Code:
    Call to 6125995555@(Ln.10000@Netborder) has failed. [ V4 192.168.1.2:5066 UDP target domain=unspecified mFlowKey=844 ] replied: 404 Not Found; from IP:192.168.1.2:5066
    
    In the Activity Log:
    Code:
    12:38:52.340|.\CallLeg.cpp(143)|Log5||CallLeg::onNewCall:[CM500002]: Info on incoming INVITE:
    INVITE sip:6125995555@192.168.1.2 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK-7658de88
    Max-Forwards: 70
    Contact: "Linksys FAX"<sip:3097@192.168.1.8:5060>
    To: <sip:6125995555@192.168.1.2>
    From: "Linksys FAX"<sip:3097@192.168.1.2>;tag=7abeaa87a43f854o0
    Call-ID: f4f140c3-ecf9f748@192.168.1.8
    CSeq: 102 INVITE
    Expires: 240
    Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
    Proxy-Authorization: Digest username="1000",realm="3CXPhoneSystem",nonce="414d535c054200bc80:947ddb1cd323c1312e555b96f46c6fc5",uri="sip:6125995555@192.168.1.2",algorithm=MD5,response="28f00585a27bd2f753bbbcb289f9edcc"
    Supported: x-sipura, replaces
    User-Agent: Linksys/SPA3102-5.1.10(GW)
    Content-Length: 0
    Remote-Party-ID: Linksys FAX <sip:3097@192.168.1.2>;screen=yes;party=calling
    
    <br>
    12:38:52.344|.\CallCtrl.cpp(346)|Log2||CallCtrl::onIncomingCall:[CM503001]: Call(569): Incoming call from Fax.3097 to <sip:6125995555@192.168.1.2><br>
    12:38:52.349|.\CallLeg.cpp(1194)|Log5||CallLeg::setRemoteSdp:Remote SDP is set for legC:569.1<br>
    12:38:52.350|.\SLServer.cpp(861)|Log5|MediaServer|MediaServerReporting::SetRemoteParty:[MS210000] C:569.1:Offer received. RTP connection: 192.168.1.8:16390(16391)<br>
    12:38:52.351|.\CallCtrl.cpp(529)|Log3||CallCtrl::onSelectRouteReq:[CM503010]: Making route(s) to <sip:6125995555@192.168.1.2><br>
    12:38:52.352|.\CallCtrl.cpp(708)|Log2||CallCtrl::onSelectRouteReq:[CM503004]: Call(569): Route 1: Unknown:6125995555@(Ln.10000@Netborder)@[Dev:sip:10000@192.168.1.2:5066,Dev:sip:10001@192.168.1.2:5066]<br>
    12:38:52.372|.\SLServer.cpp(861)|Log5|MediaServer|MediaServerReporting::SetRemoteParty:[MS210006] C:569.2:Offer provided. Connection(by pass mode): 192.168.1.8:16390(16391)<br>
    12:38:52.373|.\Target.cpp(441)|Log2||Target::makeOneInvite:[CM503025]: Call(569): Calling Unknown:6125995555@(Ln.10000@Netborder)@[Dev:sip:10000@192.168.1.2:5066]<br>
    12:38:52.573|.\CallLeg.cpp(326)|Log2||CallLeg::onFailure:[CM503003]: Call(569): Call to sip:6125995555@192.168.1.2:5066 has failed; Cause: 404 Not Found; from IP:192.168.1.2:5066<br>
    12:38:52.579|.\SLServer.cpp(861)|Log5|MediaServer|MediaServerReporting::SetRemoteParty:[MS210006] C:569.3:Offer provided. Connection(by pass mode): 192.168.1.8:16390(16391)<br>
    12:38:52.580|.\Target.cpp(441)|Log2||Target::makeOneInvite:[CM503025]: Call(569): Calling Unknown:6125995555@(Ln.10000@Netborder)@[Dev:sip:10001@192.168.1.2:5066]<br>
    12:38:52.773|.\CallLeg.cpp(326)|Log2||CallLeg::onFailure:[CM503003]: Call(569): Call to sip:6125995555@192.168.1.2:5066 has failed; Cause: 404 Not Found; from IP:192.168.1.2:5066<br>
    12:38:52.775|.\Call.cpp(1091)|Log2||Call::RouteFailed:[CM503016]: Call(569): Attempt to reach <sip:6125995555@192.168.1.2> failed. Reason: Not Found<br>
    12:38:52.775|.\Call.cpp(605)|Log2||Call::DoEndCall:[CM503020]: Normal call termination. Reason: Not found<br>
    12:39:03.795|.\Call.cpp(48)|Log5||??:Currently active calls [none]<br>
    
     
  4. leejor

    leejor Well-Known Member

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    Is port 5066, the one that you have assigned to the PSTN line in the 3102?

    By default it uses port 5061 for PSTN (5060 for FXS) and 3CX (used to , at least) default to using port 5062 for the trunk to a 3102, so one of them had to be changed.
    You can use 5066, if you wish, but both 3CX and the FXO in the 3102 must match.
     
  5. vgermscheid

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    I'll take another run at the 3102 settings, but here's a successful call from an IP phone if that helps narrow things down.

    Code:
    12:33:18.521|.\CallLeg.cpp(143)|Log5||CallLeg::onNewCall:[CM500002]: Info on incoming INVITE:
    INVITE sip:96125995555@192.168.1.2:5060;user=phone SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.74;branch=z9hG4bKaceb14fb5283622A
    Max-Forwards: 70
    Contact: <sip:3030@192.168.1.74>
    To: <sip:96125995555@192.168.1.2;user=phone>
    From: "Vince Germscheid"<sip:3030@192.168.1.2>;tag=6E0ED172-AAADB171
    Call-ID: 16e80f4e-fdc97a8d-a8131a1c@192.168.1.74
    CSeq: 2 INVITE
    Accept-Language: en
    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
    Proxy-Authorization: Digest username="3030",realm="3CXPhoneSystem",nonce="414d535c0541ff6e17:2b4d6b833ca09c2b5697d398b73d268d",uri="sip:96125995555@192.168.1.2:5060;user=phone",response="ee364798d66b2f2e3d78d383ca2e0609",algorithm=MD5
    Supported: 100rel, replaces
    User-Agent: PolycomSoundPointIP-SPIP_331-UA/3.3.0.1098
    Allow-Events: talk, hold, conference
    Content-Length: 0
    
    <br>
    12:33:18.526|.\CallCtrl.cpp(346)|Log2||CallCtrl::onIncomingCall:[CM503001]: Call(567): Incoming call from Ext.3030 to <sip:96125995555@192.168.1.2;user=phone><br>
    12:33:18.531|.\Extension.cpp(1407)|Log3||Extension::printEndpointInfo:[CM505001]: Ext.3030: Device info: Device Identified: [Man: Polycom;Mod: SoundPoint IP Series;Rev: General] Capabilities:[reinvite, replaces, unable-no-sdp, no-recvonly] UserAgent: [PolycomSoundPointIP-SPIP_331-UA/3.3.0.1098] PBX contact: [sip:3030@192.168.1.2:5060]<br>
    12:33:18.531|.\CallLeg.cpp(1194)|Log5||CallLeg::setRemoteSdp:Remote SDP is set for legC:567.1<br>
    12:33:18.532|.\SLServer.cpp(861)|Log5|MediaServer|MediaServerReporting::SetRemoteParty:[MS210000] C:567.1:Offer received. RTP connection: 192.168.1.74:2226(2227)<br>
    12:33:18.532|.\CallCtrl.cpp(529)|Log3||CallCtrl::onSelectRouteReq:[CM503010]: Making route(s) to <sip:96125995555@192.168.1.2;user=phone><br>
    12:33:18.533|.\CallCtrl.cpp(708)|Log2||CallCtrl::onSelectRouteReq:[CM503004]: Call(567): Route 1: Unknown:6125995555@(Ln.10000@Netborder)@[Dev:sip:10000@192.168.1.2:5066,Dev:sip:10001@192.168.1.2:5066]<br>
    12:33:18.577|.\SLServer.cpp(861)|Log5|MediaServer|MediaServerReporting::SetRemoteParty:[MS210002] C:567.2:Offer provided. Connection(transcoding mode): 192.168.1.2:7422(7423)<br>
    12:33:18.578|.\Target.cpp(441)|Log2||Target::makeOneInvite:[CM503025]: Call(567): Calling Unknown:6125995555@(Ln.10000@Netborder)@[Dev:sip:10000@192.168.1.2:5066]<br>
    12:33:22.384|.\CallLeg.cpp(315)|Log3||CallLeg::onAnswer:[CM503002]: Call(567): Alerting sip:10000@192.168.1.2:5066<br>
    12:33:22.385|.\CallLeg.cpp(1194)|Log5||CallLeg::setRemoteSdp:Remote SDP is set for legC:567.2<br>
    12:33:22.387|.\SLServer.cpp(861)|Log5|MediaServer|MediaServerReporting::SetRemoteParty:[MS210001] C:567.2:Answer received. RTP connection[unsecure]: 192.168.1.2:15024(15025)<br>
    12:33:22.387|.\SLServer.cpp(861)|Log5|MediaServer|MediaServerReporting::SetRemoteParty:[MS210003] C:567.1:Answer provided. Connection(transcoding mode[unsecure]):192.168.1.2:7420(7421)<br>
    12:33:28.880|.\Call.cpp(1396)|Log2||Call::Terminate:[CM503008]: Call(567): Call is terminated<br>
    
     
  6. leejor

    leejor Well-Known Member

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    Given that a call from a set works, I have to assume that the ports are set up OK. That means that there is something else that is causing the 3102 to send the 404 not found message. (T38?)
    If you set the 3CX log to verbose, there may be some additional information in there that, when comparing a working call to a non-working call, may be of help.

    You could also set up a syslog server and make use of the logs that can be set to come from the 3102. That may tell you what is "not found". You have to run something like http://www.solarwinds.com/register/kiwi_registration.aspx?Program=876&c=70150000000Es8J

    It's now a free trial but should allow you to see what's going on before it expires. You have to run that on another PC on the network and enable the syslog on the 3102 (System tab/optional Network Configuration---Syslog server/ Debug Server/ Debug level).

    Keep in mind that getting a Fax machine to work through 3CX may turn out to be a bit of a headache, as a number of people have found, you may be better off connecting it to the FXS port of the 3102 for outbound calls. see what you come up with in the logs.

    Did you see the second line in the 3CX link in your first post "This device does NOT support T38 fax transmission on the Gateway Port and therefore cannot be used to connect to an analog line to receive or send faxes with 3CX PhoneSystem."
     
  7. vgermscheid

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    In looking back at the two calls, I notice that the invite is different, on the one that works it's "INVITE sip:96125995555@192.168.1.2:5060;user=phone SIP/2.0", and the one that doesent is "INVITE sip:6125995555@192.168.1.2 SIP/2.0". Would this difference account for the call not going through and, if so, how do I get it to call on port 5060?
     
  8. leejor

    leejor Well-Known Member

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    The call is making it's way through 3CX and then continuing on to the 3102 trunk, that is where it is failing.

    Did you not see the warning (on the 3CX page, regarding the 3102) about FAX not working over a 3102?

    Read, the first paragraph, again, carefully, and then decide if you want to continue...
    http: //www.3cx.com/voip-gateways/linksys-3102/ - NO LONGER AVAILABLE

    There are some PSTN Gateways out there that do support T.38 if you do a search.
     
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