New install - 1 line 1 Endpoint

Discussion in '3CX Phone System - General' started by ZenMasta, Sep 1, 2010.

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  1. ZenMasta

    ZenMasta New Member

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    Just installed v9. And I'm trying to to setup a pots line and 1 softphone.

    Problem:
    • When I call in (from my cell etc) it rings once or twice and then all I hear is a dial tone sound.
    • I can't dial anything without a forbidden message and a couple of beeps. Doesn't matter if it is an extension or an outside number.

    I'm using a Grandstream GXW-4104 gateway. I have a single line connected into channel 1 on that device. I was trying to follow this tutorial http:/ /www.3cx.com/voip-gateways/Grandstream-GXW-41044108/ NO LONGER AVAILABLE
    Gateway:
    http://buggyonpurpose.com/random/SS-201 ... .31.20.png Status Tab
    http://buggyonpurpose.com/random/SS-201 ... .35.31.png Channels Tab
    http://buggyonpurpose.com/random/SS-201 ... .36.42.png Dial Plan
    http://buggyonpurpose.com/random/SS-201 ... .37.11.png Profile 1

    3CX:
    http://buggyonpurpose.com/random/SS-201 ... .39.18.png Trunk Status
    http://buggyonpurpose.com/random/SS-201 ... .41.02.png Extension Status

    Softphone:
    http://buggyonpurpose.com/random/3CXPho ... .54.07.png Forbidden

    Server Activity Log:
    Code:
    12:00:17.375  Requests rate from IP 127.0.0.1:1149 is too high! Blacklisted for 334 seconds
    12:00:17.375  [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
      INVITE sip:*999@localhost:5060 SIP/2.0
      Via: SIP/2.0/UDP 127.0.0.1:1149;branch=z9hG4bK-d8754z-5a61e377cc480454-1---d8754z-;rport=1149
      Max-Forwards: 70
      Contact: <sip:100@127.0.0.1:1149;rinstance=157924ea8ffde435>
      To: <sip:*999@localhost:5060>
      From: <sip:100@localhost:5060>;tag=e71e1347
      Call-ID: ODY3Y2Y0NGM4YzQ4NmI2M2Y2OGQ0MTg3ZGU5NzhlMmE.
      CSeq: 1 INVITE
      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
      Supported: replaces
      User-Agent: 3CXPhone 4.0.13679.0
      Content-Length: 0
      
    12:00:17.375  [CM302001]: Authorization system can not identify source of: SipReq:  INVITE *999@localhost:5060 tid=5a61e377cc480454 cseq=INVITE contact=100@127.0.0.1:1149 / 1 from(wire)
    11:59:07.265  [CM306003]: SIP IP:port mapping (99.59.241.22:57100) resolved by STUN server 96.9.132.83:3478 differs from the one (99.59.241.22:64404 resolved by STUN server 96.9.132.79
    11:59:07.015  [CM506003]: Resolved SIP external IP:port has changed to (99.59.241.22:57100) on Transport 192.168.13.163:5060
    11:58:11.343  [CM302001]: Authorization system can not identify source of: SipReq:  SUBSCRIBE 100@localhost:5060 tid=845f13688104353b cseq=SUBSCRIBE contact=100@127.0.0.1:1149 / 1 from(wire)
    11:54:06.234  Requests rate from IP 127.0.0.1:4972 is too high! Blacklisted for 334 seconds
      INVITE sip:*999@localhost:5060 SIP/2.0
      Max-Forwards: 70
      To: <sip:*999@localhost:5060>
      Call-ID: MjU2OGRhYmUwYjVjMGY0NTZiMDA4OTJkOTIxMmE5NDY.
      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
      User-Agent: 3CXPhone 4.0.13679.0
      
    11:54:03.125  [CM302001]: Authorization system can not identify source of: SipReq:  SUBSCRIBE 100@localhost:5060 tid=6f780c45a6171502 cseq=SUBSCRIBE contact=100@127.0.0.1:4972 / 1 from(wire)
    11:52:07.218  Requests rate from IP 127.0.0.1:4045 is too high! Blacklisted for 334 seconds
      INVITE sip:2@localhost:5060 SIP/2.0
      Max-Forwards: 70
      To: <sip:2@localhost:5060>
    
    I only see server activity when trying to dial from the soft phone. If I try to call the line I don't see any activity.

    Thanks.
     
  2. leejor

    leejor Well-Known Member

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    Did you give the Grandstream a fixed IP address that fits with your LAN IP range? Did you use the same IP in the trunk datafill in 3CX?
     
  3. ZenMasta

    ZenMasta New Member

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    I wont beat around the bush... I'm a little new to this so the lingo is hard new to me. I think I've configured and so I posted all of those screen grabs to show what it looks like on my end..
     
  4. leejor

    leejor Well-Known Member

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    The PC running 3CX and the Grandstream should each ave a fixed IP, one you assigned. Something like 192.168.1.XX, with the XX being different for each device on your network. That is just an example, it depends on what range of numbers your router is set to "deal" with. I don't see a 192.168.xx.xx IP in your logs. Most (not all) private consumer networks will use 192.168.XX.XX. Go over the settings again.
     
  5. ZenMasta

    ZenMasta New Member

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    Server and soft phone are on the same computer, softphone is configured to use localhost.

    Yes they both have their own ip and you can see each one in the screenshots.
     
  6. LeonidasG

    LeonidasG Support Team
    Staff Member 3CX Support

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    Hi,

    1) Please go to the 3CX VoIP Phone > Connections > Your Connection > In the IP Fields please remove "localhost" and put the actual IP of the PBX instead.

    2)
    - Go to your PBX > PSTN and delete the PSTN Connection you created.
    - Create a new one, select Grandstream > GX4104 > Enter the IP of the Device which is 192.168.13.156, press next until you reach the Rule Creation Page.
    - Once you have reached the Rule Creation page, in the second box where it says "Calls to numbers starting with (Prefix)" put 0 and press ok.

    Now from your VoIP Phone try calling your Mobile Phone by adding a 0 in front of the number you are trying to dial.
    If your Phone number is 99112233, then you must call 099112233.


    Try this and let us know how it goes.
     
    Stop hovering to collapse... Click to collapse... Hover to expand... Click to expand...
  7. ZenMasta

    ZenMasta New Member

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    Hi LeonidasG,
    I think maybe we're closer now.

    When I dial in from another outside line it will still only ring once or twice and then give a dial tone.

    Now if I try to dial 01234567 (my cell) it does two short rings and then says connected but my cell never rang.

    Here is the updated server log during the outbound attempt.
    Code:
    08:44:14.350  [CM503008]: Call(2): Call is terminated
    08:44:02.991  [CM503007]: Call(2): Device joined: sip:10000@192.168.13.156:5060;transport=udp
    08:44:02.975  [CM503007]: Call(2): Device joined: sip:100@127.0.0.1:1447;rinstance=4375a50612ad7c24
    08:44:02.959  [CM505002]: Gateway:[Grandstream GXW 4104] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream GXW4104 (HW 1.1, Ch:8) 1.3.1.6] PBX contact: [sip:10000@192.168.13.163:5060]
    08:44:02.959  [CM503002]: Call(2): Alerting sip:10000@192.168.13.156:5060;transport=udp
    08:44:02.913  [CM503025]: Call(2): Calling PSTNline:2542356@(Ln.10000@Grandstream GXW 4104)@[Dev:sip:10000@192.168.13.156:5060;transport=udp]
    08:44:02.881  [CM503004]: Call(2): Route 1: PSTNline:2542356@(Ln.10000@Grandstream GXW 4104)@[Dev:sip:10000@192.168.13.156:5060;transport=udp]
    08:44:02.881  [CM503010]: Making route(s) to <sip:02542356@192.168.13.163:5060>
    08:44:02.866  [CM505001]: Ext.100: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXPhone 4.0.13679.0] PBX contact: [sip:100@127.0.0.1:5060]
    08:44:02.866  [CM503001]: Call(2): Incoming call from Ext.100 to <sip:02542356@192.168.13.163:5060>
    
    As before, there is not updated server log if I try to call in.

    One more question, slightly off topic here. When I press the clear button in the server activity log, the log is cleared but only temporarily. If I click to view another pbx module and then come back to the log all the activity is still there. And then if I clear the log try to dial out through the soft phone and hit refresh, it will not refresh.
     
  8. leejor

    leejor Well-Known Member

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    It sounds like the outgoing route (gateway) is set for two stage, rather than one stage dialling. That would explain the dialtone you are getting.
     
  9. ZenMasta

    ZenMasta New Member

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    I would have guessed regardless of the gateway being 1 or 2 stage, wouldn't affect inbound calls.

    However I changed it to single stage per your suggestion and it didn't make any difference on inbound calls.

    It did however make a difference on outbound calls (dialing from 3cx phone).
    Here is a new excerpt from the server log. Call 6 is when I had set the gateway to stage 1, and call 7 is when I set it back to stage 2.

    As I mentioned previously when I make an outgoing call the phone will ring and then say connected. (Stage 2)
    When I changed it to stage 1 per your suggestion. I get a few rings and then eventually it says server failure. (Stage 1)

    And again, if I try to call the pbx from my cell or another non pbx phone I don't see anything in the server activity log.
     
  10. leejor

    leejor Well-Known Member

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    The reason i suggested it may be the a two stage dialling issue is that you said you got dialtone, after ringback, when calling in from an outside line. So, the way I'm picturing it, is, call coming in on a port on the gateway and then being forwarded back out on another gateway port to your mobile or another outside number. If you are getting dialtone, then i was suspecting that the call was coming in and the receiving dialtone from the outbound post on the gateway, You might get that when set for two stage dialling.

    Gateways are generally set for one stage dialling as you want digits sent to it then dialled out on the PSTN line. Two stage dialling allows you to get dialtone and then dial digits that may or may not be routed/permitted/barred based on a dialplan in the gateway.

    You are saying that there isn't even a log on 3CX showing an inbound call, I would think that you need to resolve that problem before going any further. Do a search of the forums for info on the 4104, you aren't the first person to try to set on up with 3Cx, or have problems. I'm assuming that you using more that one POTS line with your gateway.
     
  11. ZenMasta

    ZenMasta New Member

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    This whole installation is just for me to get a feel for 3cx and also the gateway to see if sound quality sounds normal (same as our pots phones) because I'm not excited about buying an $800 sangoma card.

    So in my situation. I have 1 computer, 1 soft phone, 1x4 port gateway, but I am only using one analog line.

    When I say I am calling in or calling from an outside line (hopefully I'm getting the terminology right). I mean I'm just using a pots line/cell phone that is no way connected to 3cx/gateway.

    Ultimately Yes I will hope to have 5 end points and 4 analog lines if/when we buy a mini license. But for now I'm just trying to test with the bare minimum.
     
  12. ZenMasta

    ZenMasta New Member

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    The problem at the moment is that I cannot make an outgoing call.

    I will hear a couple of rings and then it will connect... yet the number that was dialed doesn't ring so I'm confused by this.
    I've attached screens of all the configuration pages of the gateway.

    I've tried these recent settings too without any affect

    FXO Lines:
    FXO Termination:
    1. Enable current disconnect: ch1-4:N;
    2. Enable Tone Disconnect: ch1-4:Y;

    Channel Dialing to PSTN:
    1. Wait for dial tone ch1-4:N;
    3. Min Delay Before Dial PSTN Number: ch1-4:750;

    Channels:
    Channel Specific Settings:
    1. DTMF Methods1-7: ch1-4:1

    Any suggestions?
     

    Attached Files:

  13. ZenMasta

    ZenMasta New Member

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    bump still need help with this.
     
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