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PSTN Gateway Showing As Phone

Discussion in '3CX Phone System - General' started by bluemidnightrunner, Nov 27, 2013.

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  1. bluemidnightrunner

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    Guys,

    First post with this excellent bit of software and there is loads of help with guides etc.

    I'm having a bit of an issue with a Patton PSTN Gateway in that after the config file has been uploaded, in the 3CX console it is showing in the phones tab rather than the Gateway tab.

    Is this normal?
    The Yealink Phone and 3CX soft phone have all appeared in the same which would seem to be correct.

    Thanks in advance for the help
     
  2. mlump

    mlump New Member

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    What Patton Gateway are you using and what are you using it for. Is it an FXS Gateway to register analog lines? If so that behavior is correct. If its not and its meant for the use of Incoming Analog Lines then you have a problem. If that's the case Re Run the PSTN setup wizard and make sure you select the template for FXO.
     
  3. bluemidnightrunner

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    Hi it's a Patton SmartNode 4112 and I'm using for Analogue line to the PBX and vice versa.
     
  4. bluemidnightrunner

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    OK, got rather a lot further now and it would seem that I had to re-download the template for the device and it's now gone through the correct wizard.

    However :eek:)

    The Trunks are not registering anywhere......
     
  5. mlump

    mlump New Member

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    Go to the VoIP/PSTN Gateways and select add Gateway, from there go to the list of Patton Gateways. Does It list "SN-4112 2-port FXO - R6.x"? If it does not list this and instead lists only "SN-4112 Analog 1xETH 2xFXS - R6.x" then you need to go to 3CX Phone system Updates -> VoIP Gateway Template Updates and select "Patton SN4112 2-port FXO" and click download. Once done you should have the "SN-4112 2-port FXO - R6.x" gateway listed under the add Gateway Wizard.

    Note: When you go to download the "Patton SN4112 2-port FXO" template under updates it may already show as up to date but once you re-download it you should have the proper template to complete your setup.
     
  6. mlump

    mlump New Member

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    You May want to Factory reset your Patton Gateway then re-upload the new config afterwards. There may be some left overs from the FXS setup getting in the way. I have used that template before without issue.
     
  7. bluemidnightrunner

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    Well I seem to have got a step further but still cannot make or receive calls. This is what the log shows:
    28-Nov-2013 13:47:22.213 Leg L:9.1[Extn] is terminated: Cause: BYE from PBX
    28-Nov-2013 13:47:22.171 [CM503020]: Call(C:9): Normal call termination. Call originator: Extn:100. Reason: Not available
    28-Nov-2013 13:47:22.171 L:9.1[Extn] failed to reach Out#:>>Rule{Rule for WPPT}>>079xxxxxx, reason Temporarily Unavailable
    28-Nov-2013 13:47:22.170 [CM503026]: Call(C:9): Route 1 to PSTNlineWPPT is not active (Busy/Not registered)
    28-Nov-2013 13:47:22.170 Call(C:9): from L:9.1[Extn]: There are no available outbound lines on Trunk 'PSTNlineWPPT' at this time. First line: 10000
    28-Nov-2013 13:47:22.170 [CM303003]: There are no available outbound lines on gateway WPPT at this time. First line: Lc:10000(@WPPT[<sip:10000@192.168.254.11:5060>])
    28-Nov-2013 13:47:22.170 Line limit check: Current # of calls for line Lc:10001(@WPPT[<sip:10001@192.168.254.11:5060>]) is 0; limit is 1
    28-Nov-2013 13:47:22.170 Line limit check: Current # of calls for line Lc:10000(@WPPT[<sip:10000@192.168.254.11:5060>]) is 0; limit is 1
    28-Nov-2013 13:47:22.170 Call(C:9): Call from Extn:100 to 07xxxxxxx matches outbound rule 'Rule for WPPT'

    On the gateway the light for the 0/0 voice port is lit up but the Voip Line is not. Don't know if this makes any difference.
    Interestingly enough when a call is inbound there is nothing in the log and there doesn't seem to be anything happening on the gateway either.
     
  8. bluemidnightrunner

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    Again another step forward.

    Reset the Gateway and uploaded the config BUT! didn't press save this time and the correct config was applied and I now have registered trunks. Yay!

    However :0( it would seem that there is now another error showing that no lines are available or all are busy. I'll grab the log file and show.

    Thanks again guys for all the help
     
  9. bluemidnightrunner

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    OK, everything looks to be right now but I still get an error. Log below:

    29-Nov-2013 07:26:21.509 Leg L:20.1[Extn] is terminated: Cause: BYE from PBX
    29-Nov-2013 07:26:21.507 [CM503020]: Call(C:20): Normal call termination. Call originator: Extn:100. Reason: Busy
    29-Nov-2013 07:26:21.507 Leg L:20.3[Line:10000>>079xxxxx] is terminated: Cause: 600 Busy Everywhere/INVITE from 192.168.254.11:5062
    29-Nov-2013 07:26:21.507 L:20.1[Extn] failed to reach Line:10000>>07948302612, reason Busy
    29-Nov-2013 07:26:21.507 Call to T:Line:10000>>07xxxxxxx@[Dev:sip:10001@192.168.254.11:5062] from L:20.1[Extn] failed, cause: Cause: 600 Busy Everywhere/INVITE from 192.168.254.11:5062
    29-Nov-2013 07:26:21.506 [CM503003]: Call(C:20): Call to <sip:079xxxxxx@192.168.254.11:5062> has failed; Cause: 600 Busy Everywhere/INVITE from 192.168.254.11:5062
    29-Nov-2013 07:26:21.317 Leg L:21.1[Line:10001<<10001] is terminated: Cause: BYE from PBX
    29-Nov-2013 07:26:21.307 [CM503020]: Call(C:21): Normal call termination. Call originator: Line:10001<<10001. Reason: Busy Everywhere
    29-Nov-2013 07:26:21.307 [CM503019]: Call(C:21): Rejected! Line is 'Busy' because number of simultaneous calls per line Line:10001<<10001 is limited by Administrator
    29-Nov-2013 07:26:21.307 Call(C:21): is rejected. Incoming INVITE: Invite-IN Recv Req INVITE from 192.168.254.11:5062 tid=b77e53fe6be8f2611 Call-ID=3ab85c539b003465:
    INVITE sip:10001@192.168.254.1:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.254.11:5062;branch=z9hG4bKb77e53fe6be8f2611
    Max-Forwards: 70
    Contact: <sip:10001@192.168.254.11:5062;transport=udp>
    To: <sip:07948302612@192.168.254.1:5060>
    From: "Katrina"<sip:10001@192.168.254.11:5062>;tag=a6744c1c0c
    Call-ID: 3ab85c539b003465
    CSeq: 24852 INVITE
    Content-Type: application/sdp
    Proxy-Authorization: Digest username="10001",realm="3CXPhoneSystem",nonce="414d535c08a8d29d08:35eb13997eb9326feb49c22b9f5ae109",uri="sip:10001@192.168.254.1:5060",response="b2eb0408361d27e8c9707acc36d734a4",algorithm=MD5
    Supported: replaces
    User-Agent: Patton SN4112 JS EUI 00A0BA09A44C R6.4 2013-09-05 H323 SIP FXS FXO M5T SIP Stack/4.1.12.18
    Content-Length: 223

    v=0
    o=MxSIP 0 40 IN IP4 192.168.254.11
    s=SIP Call
    c=IN IP4 192.168.254.11
    t=0 0
    m=audio 4922 RTP/AVP 0 8 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=sendrecv
    29-Nov-2013 07:26:21.307 Line limit check: Current # of calls for line Lc:10001(@WPPT[<sip:10001@192.168.254.11:5062>]) is 2; limit is 1
    29-Nov-2013 07:26:21.302 [CM503012]: Inbound office hours rule (unnamed) for 10001 forwards to DN:100
    29-Nov-2013 07:26:20.946 Leg L:20.2[Line:10000>>07xxxxxx] is terminated: Cause: 404 Not Found/INVITE from 192.168.254.11:5060
    29-Nov-2013 07:26:20.946 [CM503025]: Call(C:20): Calling T:Line:10000>>07xxxxxx@[Dev:sip:10001@192.168.254.11:5062] for L:20.1[Extn]
     
  10. bluemidnightrunner

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    Which port should I putting the phone line in to? I've only been trying on 0/0 but from the logs it looks the system is trying on trunk 10001 which I figure should relate to port 0/1???

    Any help would really be appreciated guys
     
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