GrandStream 4108 Outbound Call not work

Discussion in '3CX Phone System - General' started by itmnetworks, Jun 1, 2010.

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  1. itmnetworks

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    Hello,

    I have GrandStream 4108 and 3cx and one PSTN Line.

    I setup my 4108 using DTMF to use on Brazil ( My Country ).

    I receive all calls, but not create outbound calls, pabx delvery for me busy signal

    If i use Stage Method = 2 and use phone number example: 991 40629422 the ippabx show for me dial tone and i need type again 40629422 to complete my call

    if i use Stage Method = 1 and use phone number example: 991 40629422 the ippabx show buse tone for me.

    please see my logs using Stage Method = 1 :

    14:56:25.033 [CM503020]: Normal call termination. Reason: Forbidden
    14:56:25.033 [CM503016]: Call(107): Attempt to reach <sip:99140629422@192.168.4.91> failed. Reason: Forbidden
    14:56:25.033 [CM503003]: Call(107): Call to sip:40629422@192.168.4.51:5060 has failed; Cause: 403 ; from IP:192.168.4.51:5060
    14:56:25.001 [CM503025]: Call(107): Calling PSTNline:40629422@(Ln.10012@GrandStream4108)@[Dev:sip:10012@192.168.4.51:5060;transport=udp]
    14:56:25.001 [MS210002] C:107.2:Offer provided. Connection(transcoding mode): 192.168.4.91:7400(7401)
    14:56:24.955 [CM503004]: Call(107): Route 1: PSTNline:40629422@(Ln.10012@GrandStream4108)@[Dev:sip:10012@192.168.4.51:5060;transport=udp]
    14:56:24.955 [MS210000] C:107.1:Offer received. RTP connection: 192.168.4.62:30146(30147)
    14:56:24.955 [CM503010]: Making route(s) to <sip:99140629422@192.168.4.91>
    14:56:24.955 Remote SDP is set for legC:107.1
    14:56:24.955 [CM505001]: Ext.1002: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Cisco-CP7960G/8.0] PBX contact: [sip:1002@192.168.4.91:5060]
    14:56:24.955 [CM503001]: Call(107): Incoming call from Ext.1002 to <sip:99140629422@192.168.4.91>
    14:56:24.955 [CM500002]: Info on incoming INVITE:
    INVITE sip:99140629422@192.168.4.91 SIP/2.0
    Via: SIP/2.0/UDP 192.168.4.62:5060;branch=z9hG4bK7d68c194
    Max-Forwards: 70
    Contact: <sip:1002@192.168.4.62:5060;transport=udp>
    To: <sip:99140629422@192.168.4.91>
    From: "1002"<sip:1002@192.168.4.91>;tag=001aa1c6d183003722e0b2d9-0bb7450d
    Call-ID: 001aa1c6-d1830012-04a2a68a-4b86d80c@192.168.4.62
    CSeq: 102 INVITE
    Expires: 180
    Accept: application/sdp
    Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, UPDATE
    Proxy-Authorization: Digest username="1002",realm="3CXPhoneSystem",uri="sip:99140629422@192.168.4.91",response="d9ef7880b19642971cc00b7a1953867e",nonce="414d535c0215dac875:23c609e189728d13ec02e07e512b5fe3",algorithm=MD5
    Supported: replaces, join, norefersub
    User-Agent: Cisco-CP7960G/8.0
    Content-Length: 0
    Remote-Party-ID: "1002" <sip:1002@192.168.4.91>;party=calling;id-type=subscriber;privacy=off;screen=yes


    My 3cx is: 3CXPSSB

    On my GrandStream 4108 i have:

    Product Model: GXW4108
    Software Version: Program-- 1.3.1.6 Loader-- 1.1.3.4 Boot-- 1.1.3.2

    FXO Lines Menu:

    Wait for Dial-Tone(Y/N): ch1-8:N;
    Stage Method(1/2): ch1-8:1;
    Unconditional Call Forward to VOIP: UserID: ch1-8:10012+; SipSever: ch1-8:p1; Sip Destination Port: ch1-8:5060++;

    Channels Menu:

    Dial Tone, RingBack Tone, Busy Tone and Reorder Tone = Brazil DTMF parameters
    DTMF Methods: ch1-8:2; ( RFC2833 )
    Local SIP Listen Port: ch1-8:5060++;
    Round-robin and/or Flexible: rr:1-8;
    Prefix to Specify Port(1 stage dialing method): 99

    Dial Plan Menu:
    PSTN Outgoing Call Dial Plan: { x+ }

    On 3CX i create Outbound Rule:

    Calls to numbers starting with (Prefix): 991
    Route 1 = GrandStream 4108
    Strip Digits = 3
    Prepend =

    I have one separate network and NIC to all IPPHONES and Gateway and 3cx ( i use 192.168.4.x isolated for voip on 3com switch )

    How can resolve this ?

    Please HELP me :)


    Best Regards,
    Rodrigo Araujo
    ITMNETWORKS
     
  2. leejor

    leejor Well-Known Member

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    You really want to be using one stage dialling. I would go over the settings in the 4108 again. Is there a log in the 4108 that would indicate why it is forbidding the call?

    According to http: //www.3cx.com/voip-gateways/Grandstream-GXW-41044108/ NO LONGER AVAILABLE

    the Wait for Dialtone should be set to "Yes". Don't know if that will make a difference.
     
  3. itmnetworks

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    Hello all,

    I see on my 4108 on Profile 1 SIP Server using network 192.168.2.x and the correct is 192.168.4.x :)

    Now i create calls from 3cx to outside using GrandStream 4108 + PSTN line and using Stage 1 dialing.

    Very thanks for all.

    Best regards,
    Rodrigo Araujo
    ITMNETWORKS
     
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