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Telephony integration RRS feed

  • Question

  •  

    Dear all,

     

    a brief description of the architecture. OCS2007 architecture connected with a non IP PBX Alcatel through a voice gateway (CISCO Router).

    From my MOC client I can dial out to the PSTN and receive calls no problem.

    Now the problem is as follows:

     

    from my MOC client I dial out a PSTN number (ok)

    I want to make a conference with a second PSTN number (nok)

     

    I guess the problem is related with configuration of my voice gateway.

    Below the q931 debugging

     

     17:54:45.430: ISDN Se2/0:15 Q931: TX -> SETUP pd = 8  callref = 0x0602
            Bearer Capability i = 0x8090A3
                    Standard = CCITT
                    Transfer Capability = Speech
                    Transfer Mode = Circuit
                    Transfer Rate = 64 kbit/s
            Channel ID i = 0xA98381
                    Exclusive, Channel 1
            Calling Party Number i = 0x0080, '39011xxxxxxx'
                    Plan:Unknown, Type:Unknown
            Called Party Number i = 0x80, '0338xxxxxx'
                    Plan:Unknown, Type:Unknown
    *Aug  4 17:54:45.470: ISDN Se2/0:15 Q931: RX <- SETUP_ACK pd = 8  callref = 0x86
    02
            Channel ID i = 0xA98381
                    Exclusive, Channel 1
    *Aug  4 17:54:45.546: ISDN Se2/0:15 Q931: RX <- CALL_PROC pd = 8  callref = 0x86
    02
    *Aug  4 17:54:49.758: ISDN Se2/0:15 Q931: RX <- ALERTING pd = 8  callref = 0x860
    2
            Progress Ind i = 0x8488 - In-band info or appropriate now available
    *Aug  4 17:54:50.858: %PS-3-MULTFAIL: There is more than one failure with the  P
    ower System 2; please resolve problems immediately
    *Aug  4 17:54:54.626: ISDN Se2/0:15 Q931: RX <- CONNECT pd = 8  callref = 0x8602

            Date/Time i = 0x08061E0B33
    *Aug  4 17:54:54.626: ISDN Se2/0:15 Q931: TX -> CONNECT_ACK pd = 8  callref = 0x
    0602
    *Aug  4 17:55:17.546: ISDN Se2/0:15 Q931: TX -> SETUP pd = 8  callref = 0x0603
            Bearer Capability i = 0x8090A3
                    Standard = CCITT
                    Transfer Capability = Speech
                    Transfer Mode = Circuit
                    Transfer Rate = 64 kbit/s
            Channel ID i = 0xA98382
                    Exclusive, Channel 2
            Calling Party Number i = 0x0080, 'giuseppe.tristano'
                    Plan:Unknown, Type:Unknown
            Called Party Number i = 0x80, '0011xxxxx'
                    Plan:Unknown, Type:Unknown
    *Aug  4 17:55:17.582: ISDN Se2/0:15 Q931: RX <- STATUS pd = 8  callref = 0x8603
            Cause i = 0x80E46C - Invalid information element contents
            Call State i = 0x06
    *Aug  4 17:55:17.582: ISDN Se2/0:15 Q931: TX -> RELEASE pd = 8  callref = 0x0603

            Cause i = 0x80E5 - Message not compatible with call state
    *Aug  4 17:55:17.586: ISDN Se2/0:15 Q931: RX <- SETUP_ACK pd = 8  callref = 0x86
    03
            Channel ID i = 0xA98382
                    Exclusive, Channel 2
    *Aug  4 17:55:17.586: ISDN Se2/0:15 **ERROR**: Ux_BadMsg: Invalid Message for ca
    ll state 19, call id 0x8584, call ref 0x603, event 0xD
    *Aug  4 17:55:17.586: ISDN Se2/0:15 Q931: TX -> STATUS pd = 8  callref = 0x0603
            Cause i = 0x80E20D - Message not compatible with call state or not imple
    mented
            Call State i = 0x13
    *Aug  4 17:55:17.614: ISDN Se2/0:15 Q931: RX <- RELEASE_COMP pd = 8  callref = 0
    x8603
            Cause i = 0x8090 - Normal call clearing
    *Aug  4 17:55:20.858: %PS-3-MULTFAIL: There is more than one failure with the  P
    ower System 2; please resolve problems immediately
    *Aug  4 17:55:31.038: ISDN Se2/0:15 Q931: RX <- DISCONNECT pd = 8  callref = 0x8
    602
            Cause i = 0x8090 - Normal call clearing
    *Aug  4 17:55:31.038: ISDN Se2/0:15 Q931: TX -> RELEASE pd = 8  callref = 0x0602
    *Aug  4 17:55:31.066: ISDN Se2/0:15 Q931: RX <- RELEASE_COMP pd = 8  callref = 0
    x8602
            Cause i = 0x8090 - Normal call clearing

     

     

    Can anyone help?

     

    Thank you in advance

    beppe

    Monday, June 30, 2008 12:10 PM

All replies

  • Hi Giuseppe01,

     

    I dont think this is a problem with your gateway.The problem seems to be this line:

     

     

    Calling Party Number i = 0x0080, 'giuseppe.tristano'

    This has to be a number. You need to find out why this is occuring. First  make sure your client is at the following version

     

    2.0.6362.62 per the following KB article

     

    http://support.microsoft.com/kb/948120/

     

    This may not be it but is a good place to start narrowing down stuff. Another issue I have seen only with testing if you are logged on using your own credentials  and you start a phone call with a phone that resolves to you self in the AB then try to start a conference call with someone else it will fail. Not sure why. I only say this as I am persuming that you are starting a the second call to one of you own numbers from the AB which is why you get 'giuseppe.tristano' in the number field.

     

    Cheers

    Chris

     

     

    Monday, June 30, 2008 1:59 PM
  • Hello Chris,

     

    Thank you very much. I applied the communicator.msp you suggested. The problem is still there. When I make the first call the Calling Party Number is correct (my MOC assigned number) whereas when I invite a second PSTN number to joing the conference the Calling Party becomes giuseppe.tristano (which is the cause of the failure).

    Just to be sure that is nothing to do with my ABS stored number I used other numbers. Same result.....

     

    Any idea?

    beppe

     

    Monday, June 30, 2008 4:53 PM
  • One thing you could try is turning on logging in the client and see the message traffic going in and out of MOC to ensure the cleint is sending the right information and use the snooper tool to look at the SIP traffic. The snooper tool is in the OCS resource kit. There is also a voip trouble shooting guide that should have more info in there about turning logging on your client.If you go into options>general tab you will see where to turn it on. A link to download the guide is below.

     

    http://www.microsoft.com/downloads/details.aspx?FamilyID=7b490758-ef9a-4442-9f0f-a5aeb4935c46&displaylang=en

     

     

    This is a great guide to use when troubleshooting OCS voice issues.

     

    The next step is to turn on logging on your front end server to see the SIP traffic. I dont think the mediation server should be causing this issue.

     

    Just to make sure what you are seeing on your gateway is correct and the gateway isnt screwing somehting up you could run debug ccsip message  through a terminal monitor or console session. This will show the live SIP messages entering your gateway from OCS.

     

    Hope this helps.

     

    Cheers

    Chris

    Monday, June 30, 2008 5:52 PM
  • hello Chris,

     

    I tried with CCSIP nothing wrong there (apparently). In addition I have a Proof of Concept in which the conferencing works with the second call showing giuseppe.tristano. I suspect the difference can be either in the way the beam of numbers on the PBX was created or a missing update on my OCS infrastructure. As you suggested I'll start from the OCS Analizer and the other Troubleshooting tools you suggests.

    Thank you very much.

    Keep you posted

     

    beppe

    Tuesday, July 1, 2008 3:29 PM