CONFIGURATION OF GRANDSTREAM GXW4004 (IN IRELAND OR UK)

Discussion in '3CX Phone System - General' started by IanMcD, Mar 10, 2009.

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  1. IanMcD

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    We have used the 3CX PBX successfully with a Linksys SPA-3102 for a few months now. For one reason and another we have decided to move to a Grandstream GWX-4004. I am making very little progress on the configuration of this - would be grateful for some advice.

    I have followed all the steps that the 3cx configuration guide suggests.

    In 3cx I have added a PSTN line for each of the ports on the gateway (I'll only actually use one). I have added these using the generic template. I am fully up to date with the 3cx software versions.

    When I look at the grandstream I can see that all ports have registered successfully. I have installed the latest firmware, versions are:
    Program-- 1.0.1.21 Bootloader-- 1.0.0.7 Core-- 1.0.0.25 Base-- 1.0.0.76

    When I try an incoming call I get the number not recognised tone. If I disconnect the phone line and redial, I will get the ringing tone.

    When I try an outgoing call (over PSTN) I get the ringing tone on the handset, but the dialled number will not ring.

    I am assuming this something to do with my configuration of the grandstream, probably in the "call progress tones" section(?). I have tried various settings here that I have found on the web including the ones shown in the attached screenshot. I tried these with and without the ch1-4: prefix.

    Any help much appreciated

    Ian
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    Server status log for outgoing call is below:

    14:51:52.491 Call::Terminate [CM503008]: Call(26): Call is terminated
    14:51:26.303 Line::printEndpointInfo [CM505002]: Gateway:[GRANDSTREAM] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream GXW-4004 V1.3A 1.0.1.21] Transport: [sip:192.168.176.115:5060]
    14:51:26.303 CallCtrl::eek:nAnsweredCall [CM503002]: Call(26): Alerting sip:10002@192.168.176.2:5060
    14:51:26.178 MediaServerReporting::SetRemoteParty [MS210002] C:26.2:Offer provided. Connection(transcoding mode): 192.168.176.115:7448(7449)
    14:51:26.116 MediaServerReporting::SetRemoteParty [MS210000] C:26.1:Offer received. RTP connection: 192.168.176.189:10142(10143)
    14:51:26.116 CallCtrl::eek:nSelectRouteReq [CM503004]: Call(26): Calling: PSTNline:10002@[Dev:sip:10002@192.168.176.2:5060, Dev:sip:10003@192.168.176.2:5060, Dev:sip:10004@192.168.176.2:5060, Dev:sip:10005@192.168.176.2:5060]
    14:51:26.053 CallCtrl::eek:nSelectRouteReq [CM503010]: Making route(s) to [sip:80823809999@192.168.176.115;user=phone]
    14:51:26.053 CallLeg::setRemoteSdp Remote SDP is set for legC:26.1
    14:51:26.053 Extension::printEndpointInfo [CM505001]: Ext.101: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [elmegIP290-3.4] Transport: [sip:192.168.176.115:5060]
    14:51:26.037 CallCtrl::eek:nIncomingCall [CM503001]: Call(26): Incoming call from Ext.101 to [sip:80823809999@192.168.176.115;user=phone]
    14:51:25.975 CallLeg::eek:nNewCall [CM500002]: Info on incoming INVITE:
    INVITE sip:80823809999@192.168.176.115;user=phone SIP/2.0
    Via: SIP/2.0/UDP 192.168.176.189:5060;branch=z9hG4bK-1hwo6nshorgy;rport=5060
    Max-Forwards: 70
    Contact: [sip:101@192.168.176.189:5060;line=ga88nc7d]
    To: [sip:80823809999@192.168.176.115;user=phone]
    From: "101"[sip:101@192.168.176.115];tag=n3ttyqckgp
    Call-ID: 3c54dd35ef42-7zcumpn47kb1@192-168-176-189
    CSeq: 2 INVITE
    Session-Expires: 3600
    Accept: application/sdp
    Accept-Language: en
    Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
    Proxy-Authorization: Digest username="101",realm="3CXPhoneSystem",nonce="12881170285:3695341f8d6448b1fe0e97498f65ade9",uri="sip:80823809999@192.168.176.115;user=phone",response="9385fbee054e74f93393b75ec9f97fdd",algorithm=md5
    Supported: timer, 100rel, replaces
    User-Agent: elmegIP290-3.4
    Allow-Events: talk, hold, refer
    Content-Length: 0
    P-Key-Flags: keys="3"


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    I have attached the settings I currently have for the tones
     

    Attached Files:

  2. William400

    William400 Well-Known Member

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    Hi

    Can you please try these out and let us know.

    # Dial Tone
    ch1-8:f1=350@-13,f2=440@-13,c=0/0;
    #### Ringback Tone
    ch1-8:f1=440@-11,f2=480@-11,c=200/400;
    # Busy Tone
    ch1-8:f1=400@-10,f2=0,c=375/375;
    # Reorder Tone
    ch1-8:f1=400@-10,f2=0,c=40/35-225/525;
    # Confirmation Tone
    ch1-8:f1=1400@-10,f2=0,c=20/0;
     
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  3. IanMcD

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    William,

    Thanks for your reply, but I now think I have the wrong device. When I bought the GWX4004 I thought I was getting a device that would sit between 3CX and our PSTN line. i.e. similar to the SPA-3102. The documentation with the GWX4004 suggested otherwise, but I persevered because I knew the device was mentioned on the 3CX website. From what I can see the GWX4004 could be used to interface conventional phones with VOIP functionality (not what I'm after). I plan to send this back and get a GWX4104.
     
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