Voipstunt + OKI ATA troubles

Discussion in 'UK VOIP' started by Jan De Luyck, Jun 27, 2006.

  1. Jan De Luyck

    Jan De Luyck Guest

    Hello all,

    I've got a subscription with voipstunt, and I've been using it for three months
    without any troubles using my OKI BMG7012.

    Since the 17th tho, I am unable to place any call using my ATA adapter. Using
    a softphone (twinkle on linux) works perfectly, so there are no troubles in my
    home setup.

    I've traced the outgoing sip conversation, and as I dial out a number, I get an
    401 Unauthorized back from the voipstunt and I get a busy tone from the ATA.

    Any ideas what I could check? I've verified all the settings (I've re-entered
    them this morning after doing a full reset), the internet connection is okay
    (normally it's hooked up behind a wifi router, but I connected it straight on
    the net this morning too), so I'm a bit stumped what to do next.

    Voipstunt themselves "don't officially support SIP", and they don't respond to
    mails - atleast sofar not.

    Thanks,

    Jan

    --
    Linux rutabaga 2.6.17 #1 PREEMPT Mon Jun 19 08:02:42 CEST 2006 i686 GNU/Linux
    "To IBM, 'open' means there is a modicum of interoperability among some of their
    equipment."
    -- Harv Masterson
    Jan De Luyck, Jun 27, 2006
    #1
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  2. Jan De Luyck

    Jon Farmer Guest

    Jan De Luyck wrote:
    > Hello all,
    >
    > I've got a subscription with voipstunt, and I've been using it for three months
    > without any troubles using my OKI BMG7012.
    >
    > Since the 17th tho, I am unable to place any call using my ATA adapter. Using
    > a softphone (twinkle on linux) works perfectly, so there are no troubles in my
    > home setup.
    >
    > I've traced the outgoing sip conversation, and as I dial out a number, I get an
    > 401 Unauthorized back from the voipstunt and I get a busy tone from the ATA.


    Can you post the SIP messages here so we can see what is going on?


    > Voipstunt themselves "don't officially support SIP", and they don't respond to
    > mails - atleast sofar not.


    Never used Voipstunt myself but if what you say above is true it turns
    me right off them.


    Regards

    Jon
    Jon Farmer, Jun 28, 2006
    #2
    1. Advertising

  3. Jan De Luyck

    Jan De Luyck Guest

    On 2006-06-28, Jon Farmer wrote:
    > Jan De Luyck wrote:
    >> Hello all,
    >>
    >> I've got a subscription with voipstunt, and I've been using it for three
    >> months without any troubles using my OKI BMG7012.
    >>
    >> Since the 17th tho, I am unable to place any call using my ATA adapter. Using
    >> a softphone (twinkle on linux) works perfectly, so there are no troubles in
    >> my home setup.
    >>
    >> I've traced the outgoing sip conversation, and as I dial out a number, I get
    >> an 401 Unauthorized back from the voipstunt and I get a busy tone from the
    >> ATA.

    >
    > Can you post the SIP messages here so we can see what is going on?


    Sure. Big post coming on, it's the actual attempt to dial a PSTN line. (my own)

    I've obscured my own IP, my SIP username (username) and the number I tried to
    dial.
    They're all valid.

    192.168.33.150 is the internal ip address of the ATA.

    -------------------------------------------
    ---> Send to sip.voipstunt.com:5060 at tick 36798
    REGISTER sip:voipstunt.com SIP/2.0
    Via: SIP/2.0/UDP XX.XX.XX.XX:5060
    From: voipstunt <sip:>;tag=3381863556
    To: voipstunt <sip:>
    Call-ID: 1973839901@192.168.33.150
    CSeq: 14 REGISTER
    Contact: <sip::5060>
    Authorization: Digest username="username", realm="sipdiscount.com",
    nonce="878586346", uri="sip:sip.voipstunt.com:5060", response="password",
    algorithm=MD5
    max-forwards: 70
    expires: 60
    Content-Length: 0


    <--- Recv from 194.120.0.202:5060 at tick 36808
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP XX.XX.XX.XX:5060
    From: voipstunt <sip:>;tag=3381863556
    To: voipstunt <sip:>
    Contact: sip:194.120.0.202:5060
    Call-ID: 1973839901@192.168.33.150
    CSeq: 14 REGISTER
    User-Agent: (Very nice Sip Registrar Server)
    Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS
    Content-Length: 0


    <--- Recv from 194.120.0.202:5060 at tick 36810
    SIP/2.0 200 Ok
    Via: SIP/2.0/UDP 192.168.33.150:5060
    From: voipstunt <sip:>;tag=3381863556
    To: voipstunt <sip:>
    Contact: sip:192.168.33.150:5060
    Call-ID: 1973839901@192.168.33.150
    CSeq: 14 REGISTER
    User-Agent: (Very nice Sip Registrar Server)
    Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS
    Content-Length: 0


    VOIP led status = on.
    <--- Recv from 80.239.235.201:5060 at tick 37273
    SIP/2.0 401 Unauthorized
    Via: SIP/2.0/UDP XX.XX.XX.XX:5060
    From: voipstunt <sip:>;tag=3381863556
    To: voipstunt <sip:>
    Contact: sip:80.239.235.201:5060
    Call-ID: 1973839901@192.168.33.150
    CSeq: 13 REGISTER
    User-Agent: (Very nice Sip Registrar Server)
    Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS
    WWW-Authenticate: Digest
    realm="sipdiscount.com" ,nonce="603052190" ,algorithm=MD5
    Content-Length: 0

    ---> Send to sip.voipstunt.com:5060 at tick 42608
    REGISTER sip:voipstunt.com SIP/2.0
    Via: SIP/2.0/UDP XX.XX.XX.XX:5060
    From: voipstunt <sip:>;tag=1003685425
    To: voipstunt <sip:>
    Call-ID: 1973839901@192.168.33.150
    CSeq: 16 REGISTER
    Contact: <sip::5060>
    Authorization: Digest username="username", realm="sipdiscount.com",
    nonce="4003847063", uri="sip:sip.voipstunt.com:5060", response="password",
    algorithm=MD5
    max-forwards: 70
    expires: 60
    Content-Length: 0


    string=0

    DTMF EVENT line 0: 0
    MGCP SIGNAL: endpt 0, cnx -1, evt 11 (DIALTONE) = OFF
    [VoIP -> PSTN] VoIP DTMF digit Input 0.
    processCmdQ: EPTCMD_SIGNAL
    stopTone: devid 0 keeping tone vhd for tone det
    string=0

    DTMF EVENT line 0: 0
    [VoIP -> PSTN] VoIP DTMF digit Input 0.
    <--- Recv from 194.221.62.206:5060 at tick 42697
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP XX.XX.XX.XX:5060
    From: voipstunt <sip:>;tag=1003685425
    To: voipstunt <sip:>
    Contact: sip:194.221.62.206:5060
    Call-ID: 1973839901@192.168.33.150
    CSeq: 16 REGISTER
    User-Agent: (Very nice Sip Registrar Server)
    Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS
    Content-Length: 0


    <--- Recv from 194.221.62.206:5060 at tick 42700
    SIP/2.0 200 Ok
    Via: SIP/2.0/UDP 192.168.33.150:5060
    From: voipstunt <sip:>;tag=1003685425
    To: voipstunt <sip:>
    Contact: sip:192.168.33.150:5060
    Call-ID: 1973839901@192.168.33.150
    CSeq: 16 REGISTER
    User-Agent: (Very nice Sip Registrar Server)
    Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS
    Content-Length: 0


    VOIP led status = on.
    string=5

    DTMF EVENT line 0: x
    [VoIP -> PSTN] VoIP DTMF digit Input x.
    string=3

    DTMF EVENT line 0: x
    [VoIP -> PSTN] VoIP DTMF digit Input x.
    string=3

    DTMF EVENT line 0: x
    [VoIP -> PSTN] VoIP DTMF digit Input x.
    string=3

    DTMF EVENT line 0: x
    [VoIP -> PSTN] VoIP DTMF digit Input x.
    string=3

    DTMF EVENT line 0: x
    [VoIP -> PSTN] VoIP DTMF digit Input x.
    string=3

    DTMF EVENT line 0: x
    [VoIP -> PSTN] VoIP DTMF digit Input x.
    string=4

    DTMF EVENT line 0: x
    [VoIP -> PSTN] VoIP DTMF digit Input x.
    string=1

    DTMF EVENT line 0: x
    [VoIP -> PSTN] VoIP DTMF digit Input x.
    string=0

    DTMF EVENT line 0: x
    [VoIP -> PSTN] VoIP DTMF digit Input x.
    string=0

    DTMF EVENT line 0: x
    [VoIP -> PSTN] VoIP DTMF digit Input x.
    string=8

    DTMF EVENT line 0: x
    [VoIP -> PSTN] VoIP DTMF digit Input x.
    string=6

    DTMF EVENT line 0: x
    [VoIP -> PSTN] VoIP DTMF digit Input x.
    [VoIP -> PSTN] VoIP DTMF digit Input T.
    [VoIP -> PSTN] VoIP digit string dial out.
    >>> Send to 127.0.0.1:2727 ---

    NTFY 306 aaln/1@[192.168.33.150] MGCP 1.0
    X: 10
    O: x,x,x,x,x,x,x,x,x,x,x,x,T
    >>>

    <<< Recv from 127.0.0.1:2727 ---
    200 306 OK
    <<<

    A: 9 edge down.
    ======== Alan debug: CA has Response (1)


    Parser status 200
    Tmr adj to=0, RTT=58, DEV=65, RTO=188
    <<< Recv from 127.0.0.1:2727 ---
    CRCX 3015 AALN/1@[192.168.33.150] MGCP 1.0
    C: 0
    L: p:20,
    a:G729A;PCMU;PCMA;G.723.1-5.3;G723.1;G723.1A-5.3;G723.1A;G729B;telephone-event,
    fmtp:"telephone-event 0-15", s:eek:ff
    M: recvonly
    <<<

    ======== Alan debug: CA has Response (1)


    Parser status 200
    MGCP CREATE CNX: endpt 0, cnx 0
    ccConnection remote (decode) codec list:
    (9): 18=7:0 0=0:0 8=1:0 106=9:0
    ccConnection local (encode) codec list:
    (9): 18=7:0 0=0:0 8=1:0 106=9:0
    ccConnection: codec: 7, period: 20, vad: 0, relay: 2
    Connection Mode: RecvOnly
    ccConnection: Attached device 0 to stream 0 for cxid 0
    >>> Send to 127.0.0.1:2727 ---

    200 3015 OK
    I: 5A51

    v=0
    o=Broadcom 23121 3015 IN IP4 XX.XX.XX.XX
    s=MGCP call
    c=IN IP4 XX.XX.XX.XX
    b=AS:64
    t=0 0
    m=audio 5004 RTP/AVP 18 0 8 106 4 107 108 109 110
    a=rtpmap:106 G.723.1-5.3/8000
    a=rtpmap:107 G723.1A-5.3/8000
    a=rtpmap:108 G723.1A/8000
    a=rtpmap:109 G729B/8000
    a=rtpmap:110 TELEPHONE-EVENT/8000
    a=fmtp:110 0,1,2,3,4,5,6,7,8,9,10,11,12,13,14,15
    a=recvonly
    a=ptime:20
    >>>

    processCmdQ: EPTCMD_CREATE_STREAM
    processCmdQ: EPTCMD_ATTACH_DEVICE
    streamDevAttach: netvhd NULL for stream 0
    streamDevAttach: convert tone vhd to net vhd for dev 0
    HDSP: endpt 80 mode change event op1:0 op2:0

    HDSP: endpt 80 mode change event op1:2 op2:0

    length: application/sdp 15 18
    ---> Send to sip.voipstunt.com:5060 at tick 43256
    INVITE sip: SIP/2.0
    Via: SIP/2.0/UDP XX.XX.XX.XX:5060
    From: voipstunt <sip:>;tag=3212241778
    To: <sip:>
    Call-ID: 3868338711@192.168.33.150
    CSeq: 300 INVITE
    Contact: <sip::5060>
    max-forwards: 70
    Allow: INVITE, ACK, CANCEL, BYE, REFER, NOTIFY
    Content-Type: application/sdp
    Content-Length: 167

    v=0
    o=- 0 0 IN IP4 XX.XX.XX.XX
    s=-
    c=IN IP4 XX.XX.XX.XX
    b=AS:64
    t=0 0
    m=audio 5004 RTP/AVP 18 0 8 4 110
    a=rtpmap:110 telephone-event/8000
    a=fmtp:110 0-15
    a=ptime:20

    number of message sent: 6
    <<< Recv from 127.0.0.1:2727 ---
    RQNT 3016 AALN/1@[192.168.33.150] MGCP 1.0
    X: 11
    R: hu
    <<<

    ======== Alan debug: CA has Response (1)


    Parser status 200
    >>> Send to 127.0.0.1:2727 ---

    200 3016 OK
    >>>

    hdspctrl_setEcanCtrl: Setting PxD handle of VHD0 to PXD0
    HDSP: endpt 80 mode change event op1:0 op2:0

    HDSP: endpt 80 mode change event op1:3 op2:0, stream 0

    HDSP: PVE Codec for VHD 80 encoder rate change to G.729 (Annex A) 8 kbps no VAD
    2 Frames
    addAttDev: store devId 0 to streamId 0 list
    <--- Recv from 80.239.235.200:5060 at tick 43269
    SIP/2.0 401 Unauthorized
    Via: SIP/2.0/UDP XX.XX.XX.XX:5060
    From: voipstunt <sip:>;tag=3212241778
    To: <sip:>
    Contact: sip:xxxxxxxxxxxx@80.239.235.200:5060
    Call-ID: 3868338711@192.168.33.150
    CSeq: 300 INVITE
    User-Agent: (Very nice Sip Registrar Server)
    Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS
    WWW-Authenticate: Digest
    realm="sipdiscount.com" ,nonce="360197846" ,algorithm=MD5
    Content-Length: 0


    ---> Send to sip.voipstunt.com:5060 at tick 43272
    ACK sip: SIP/2.0
    Via: SIP/2.0/UDP XX.XX.XX.XX:5060
    From: voipstunt <sip:>;tag=3212241778
    To: <sip:>
    Call-ID: 3868338711@192.168.33.150
    CSeq: 300 ACK
    Content-Length: 0


    456xx: Call emCallDrop with EndptID[0].
    <<< Recv from 127.0.0.1:2727 ---
    DLCX 3017 AALN/1@[192.168.33.150] MGCP 1.0
    C: 0
    I: 5A51
    <<<

    ======== Alan debug: CA has Response (1)


    Parser status 200
    MGCP DELETE CNX: endpt 0, cnx 0
    >>> Send to 127.0.0.1:2727 ---

    250 3017 Connection was deleted
    P: PS=0, OS=0, PR=0, OR=0, PL=0, JI=0, LA=0
    >>>

    <<< Recv from 127.0.0.1:2727 ---
    RQNT 3018 AALN/1@[192.168.33.150] MGCP 1.0
    X: 12
    R: hu, oc
    S: bz
    <<<

    ======== Alan debug: CA has Response (1)


    Parser status 200
    MGCP SIGNAL: endpt 0, cnx -1, evt 9 (BUSYTONE) = ON
    >>> Send to 127.0.0.1:2727 ---

    200 3018 OK
    >>>

    processCmdQ: EPTCMD_DETACH_DEVICE
    getAttDevNum: streamId 0 cnt = 0
    streamDevDetach: convert netvhd to tonevhd
    HDSP: endpt 80 mode change event op1:0 op2:0

    hdspEvntCb: HAPINET_INGRESSPACKET: unknown vhdHdl 80
    HDSP: endpt 80 mode change event op1:2 op2:0

    processCmdQ: EPTCMD_DELETE_STREAM
    streamDelete: no connection exists for streamId 0
    processCmdQ: EPTCMD_SIGNAL
    hdspctrl_getToneVhdp: tone vhd already connected devId 0
    HDSP: Custom tone BUSY

    CAS EVENT line 0: ONHOOK
    [VoIP -> PSTN] VoIP on-hook signal coming.
    MGCP SIGNAL: endpt 0, cnx -1, evt 9 (BUSYTONE) = OFF
    >>> Send to 127.0.0.1:2727 ---

    NTFY 307 aaln/1@[192.168.33.150] MGCP 1.0
    X: 12
    O: hu
    >>>

    <<< Recv from 127.0.0.1:2727 ---
    200 307 OK
    <<<

    ======== Alan debug: CA has Response (1)


    Parser status 200
    Tmr adj to=0, RTT=50, DEV=63, RTO=176
    <<< Recv from 127.0.0.1:2727 ---
    RQNT 3019 AALN/1@[192.168.33.150] MGCP 1.0
    X: 13
    R: hd
    S:
    <<<

    ======== Alan debug: CA has Response (1)


    Parser status 200
    >>> Send to 127.0.0.1:2727 ---

    200 3019 OK
    >>>

    processCmdQ: EPTCMD_SIGNAL
    stopTone: devid 0 keeping tone vhd for tone det
    ---> Send to sip.voipstunt.com:5060 at tick 48600
    REGISTER sip:voipstunt.com SIP/2.0
    Via: SIP/2.0/UDP XX.XX.XX.XX:5060
    From: voipstunt <sip:>;tag=2059885872
    To: voipstunt <sip:>
    Call-ID: 1973839901@192.168.33.150
    CSeq: 17 REGISTER
    Contact: <sip::5060>
    max-forwards: 70
    expires: 60
    Content-Length: 0


    <--- Recv from 194.221.62.207:5060 at tick 48605
    SIP/2.0 401 Unauthorized
    Via: SIP/2.0/UDP XX.XX.XX.XX:5060
    From: voipstunt <sip:>;tag=2059885872
    To: voipstunt <sip:>
    Contact: sip:194.221.62.207:5060
    Call-ID: 1973839901@192.168.33.150
    CSeq: 17 REGISTER
    User-Agent: (Very nice Sip Registrar Server)
    Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS
    WWW-Authenticate: Digest
    realm="sipdiscount.com" ,nonce="878704455" ,algorithm=MD5
    Content-Length: 0


    ---> Send to sip.voipstunt.com:5060 at tick 48609
    REGISTER sip:voipstunt.com SIP/2.0
    Via: SIP/2.0/UDP XX.XX.XX.XX:5060
    From: voipstunt <sip:>;tag=2059885872
    To: voipstunt <sip:>
    Call-ID: 1973839901@192.168.33.150
    CSeq: 18 REGISTER
    Contact: <sip::5060>
    Authorization: Digest username="username", realm="sipdiscount.com",
    nonce="878704455", uri="sip:sip.voipstunt.com:5060", response="password",
    algorithm=MD5
    max-forwards: 70
    expires: 60
    Content-Length: 0


    <--- Recv from 194.120.0.202:5060 at tick 48621
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP XX.XX.XX.XX:5060
    From: voipstunt <sip:>;tag=2059885872
    To: voipstunt <sip:>
    Contact: sip:194.120.0.202:5060
    Call-ID: 1973839901@192.168.33.150
    CSeq: 18 REGISTER
    User-Agent: (Very nice Sip Registrar Server)
    Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS
    Content-Length: 0


    <--- Recv from 194.120.0.202:5060 at tick 48622
    SIP/2.0 200 Ok
    Via: SIP/2.0/UDP 192.168.33.150:5060
    From: voipstunt <sip:>;tag=2059885872
    To: voipstunt <sip:>
    Contact: sip:192.168.33.150:5060
    Call-ID: 1973839901@192.168.33.150
    CSeq: 18 REGISTER
    User-Agent: (Very nice Sip Registrar Server)
    Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS
    Content-Length: 0


    VOIP led status = on.
    ----------------------------------------

    Thanks for any insights you people might give me.

    Jan
    --
    Linux rutabaga 2.6.17 #1 PREEMPT Mon Jun 19 08:02:42 CEST 2006 i686 GNU/Linux
    Q: How was Thomas J. Watson buried?
    Jan De Luyck, Jun 28, 2006
    #3
  4. Jan De Luyck

    Jon Farmer Guest

    Jan De Luyck wrote:
    > On 2006-06-28, Jon Farmer wrote:
    >> Jan De Luyck wrote:
    >>> Hello all,
    >>>
    >>> I've got a subscription with voipstunt, and I've been using it for three
    >>> months without any troubles using my OKI BMG7012.
    >>>
    >>> Since the 17th tho, I am unable to place any call using my ATA adapter. Using
    >>> a softphone (twinkle on linux) works perfectly, so there are no troubles in
    >>> my home setup.
    >>>
    >>> I've traced the outgoing sip conversation, and as I dial out a number, I get
    >>> an 401 Unauthorized back from the voipstunt and I get a busy tone from the
    >>> ATA.

    >> Can you post the SIP messages here so we can see what is going on?

    >
    > Sure. Big post coming on, it's the actual attempt to dial a PSTN line. (my own)
    >
    > I've obscured my own IP, my SIP username (username) and the number I tried to
    > dial.
    > They're all valid.


    > ---> Send to sip.voipstunt.com:5060 at tick 42608
    > REGISTER sip:voipstunt.com SIP/2.0
    > Via: SIP/2.0/UDP XX.XX.XX.XX:5060
    > From: voipstunt <sip:>;tag=1003685425
    > To: voipstunt <sip:>
    > Call-ID: 1973839901@192.168.33.150
    > CSeq: 16 REGISTER
    > Contact: <sip::5060>
    > Authorization: Digest username="username", realm="sipdiscount.com",
    > nonce="4003847063", uri="sip:sip.voipstunt.com:5060", response="password",
    > algorithm=MD5
    > max-forwards: 70
    > expires: 60
    > Content-Length: 0



    Well it looks like the device is not sending back the REGISTER message
    that the server is expecting. It is a MD5 response which means both the
    nonce and the response should be MD5 hashes. In the messages you post
    all the nonce and response are non-MD5.

    Are you sure the device you are trying to use support WWW-Authentication?


    HTH

    Regards

    Jon
    Jon Farmer, Jun 28, 2006
    #4
  5. Jan De Luyck

    Jan De Luyck Guest

    On 2006-06-28, Jon Farmer wrote:
    > Jan De Luyck wrote:
    >> ---> Send to sip.voipstunt.com:5060 at tick 42608
    >> REGISTER sip:voipstunt.com SIP/2.0
    >> Via: SIP/2.0/UDP XX.XX.XX.XX:5060
    >> From: voipstunt <sip:>;tag=1003685425
    >> To: voipstunt <sip:>
    >> Call-ID: 1973839901@192.168.33.150
    >> CSeq: 16 REGISTER
    >> Contact: <sip::5060>
    >> Authorization: Digest username="username", realm="sipdiscount.com",
    >> nonce="4003847063", uri="sip:sip.voipstunt.com:5060", response="password",
    >> algorithm=MD5
    >> max-forwards: 70
    >> expires: 60
    >> Content-Length: 0

    >
    >
    > Well it looks like the device is not sending back the REGISTER message
    > that the server is expecting. It is a MD5 response which means both the
    > nonce and the response should be MD5 hashes. In the messages you post
    > all the nonce and response are non-MD5.
    >
    > Are you sure the device you are trying to use support WWW-Authentication?


    The password is an MD5 string, the nonce I didn't change.

    I can't say I can find anything in the manual I've got about it.. nothing
    about the authentication whatsoever.

    Fun thing is, it _used_ to work. It just isn't working anymore.

    I verified, the OKI still works perfectly with eg voipfone.

    Regards,

    Jan

    --
    Linux rutabaga 2.6.17 #1 PREEMPT Mon Jun 19 08:02:42 CEST 2006 i686 GNU/Linux
    A roda adora.
    -- palĂ­ndromo
    Jan De Luyck, Jun 30, 2006
    #5
    1. Advertising

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