Fast Busy Signal RRS feed

  • Question

  • Not really sure where I would go from here.  I can dial into the number from a PSTN phone but when I use the communicator I get a fast busy signal.  I have run the validation wizard and there are currently no errors.


    Pool -


    Mediation Server

    I have a DMG 4120 that has two nic

    Communications Listening address

    Gateway Listening address


    Avaya PBX

    Friday, November 30, 2007 2:45 PM

All replies


    You need to gather logs from the OCS server and DMG Gateway. You can enabling logging on the client first as we improved logging such that we include the error data from the server that experienced the error. You can also turn on logging at the pool level and view this in snooper (OCS Resource Kit)


    The key is to find which device has the problem and the error they report.

    Friday, November 30, 2007 7:42 PM



    See Communicator call post by me for logs.  As well in the event log i get the following.



    Event Type: Warning
    Event Source: Communicator
    Event Category: None
    Event ID: 11
    Date:  12/3/2007
    Time:  9:44:54 AM
    User:  N/A
    Computer: MACHINENAME
    A SIP request made by Communicator failed in an unexpected manner (status code 80ef01e0). More information is contained in the following technical data:
     RequestUri:   sip:7033618619;phone-context=dialstring@company.com;user=phone
    From:         sip:FN.LN@company.com;tag=e3bdf93e57
    To:           sip:7033618619;phone-context=dialstring@company.com;user=phone;tag=E6AB7C102249BCCE05033FBF1C28D7C0
    Call-ID:      b2110f51e5274e9d867f60b5e9a380f6
    Content-type: application/sdp;call-type=audiovideo

    o=- 0 0 IN IP4 192.254.A.x
    c=IN IP4 192.254.A.x
    t=0 0
    m=audio 17280 RTP/AVP 114 111 112 115 116 4 8 0 97 101
    a=candidate:yQNHRaCV5b9jk56XNVeAg+nkg5Curkt8Abz+D8iOhFc 1 pAOMQRPCmvPERMgS1UBwsQ UDP 0.900 192.254.A.x 17280
    a=candidate:yQNHRaCV5b9jk56XNVeAg+nkg5Curkt8Abz+D8iOhFc 2 pAOMQRPCmvPERMgS1UBwsQ UDP 0.900 192.254.A.x 56192
    a=cryptoscale:1 client AES_CM_128_HMAC_SHA1_80 inline:tgZ70qog+RlX2ZIV2Hi4VECSrqx4xbJ5214EUWmh|2^31|1:1
    a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:IXdmRrP5I2o359WDWVurOLUDqBIZJbjtX9ySKGc3|2^31|1:1
    a=rtpmap:114 x-msrta/16000
    a=fmtp:114 bitrate=29000
    a=rtpmap:111 SIREN/16000
    a=fmtp:111 bitrate=16000
    a=rtpmap:112 G7221/16000
    a=fmtp:112 bitrate=24000
    a=rtpmap:115 x-msrta/8000
    a=fmtp:115 bitrate=11800
    a=rtpmap:116 AAL2-G726-32/8000
    a=rtpmap:4 G723/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:97 RED/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16

    Response Data:

    101  Progress Report
    ms-diagnostics:  14012;reason="Dialstring phone context used no translation done";source="servername.corp.company.com";appName="TranslationService"

    480  Temporarily Unavailable
    ms-diagnostics:  2;reason="See response code and reason phrase";source="servername.corp.company.com";AppUri="http://www.microsoft.com/LCS/DefaultRouting"

     If this error continues to occur, please contact your network administrator. The network administrator can use a tool like winerror.exe from the Windows Resource Kit or lcserror.exe from the Office Communications Server Resource Kit in order to interpret any error codes listed above.

    For more information, see Help and Support Center at http://go.microsoft.com/fwlink/events.asp.

    Monday, December 3, 2007 5:13 PM
  • I thought I replied so excuse me if it results in 2 responses -


    I do not have much detail for the error you provided so I would give some general suggestions - Did the event log from the server have any other details? Is the routing application enabled? What about the mediation server logs is there any errors their?


    I would request the sipstack component of the debug logging from the OCS Pool and also a real simple diagram of the topology as related to the components involved in getting a call to the PSTN.

    Monday, December 3, 2007 8:12 PM