Explanation of this log needed

Discussion in 'CRM / Helpdesk / App Integration' started by EdBirchall, Apr 15, 2013.

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  1. EdBirchall

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    I don't know if this is in the correct forum, but we are in debug mode and inbound calls are terminating immediately. Unfortunately, this is our first experience working with VAD or even with a phone system, so any help would be appreciated. We've found a number of similar forum subjects but none seem to match the situation we are seeing.

    At this point, we don't even know what any of this log means. if you can help, could you make the explanation as simple as possible? We are trying to get up to speed with the terminology and abbreviations, but it all takes time. Thanks in advance.

    15-Apr-2013 09:57:19.881 Leg L:43.1[Line:10002<<xxxxxxxxxx] is terminated: Cause: BYE from PBX

    15-Apr-2013 09:57:19.881 [CM503008]: Call(C:43): Call is terminated

    15-Apr-2013 09:57:19.879 Leg L:43.2[Ivr] is terminated: Cause: BYE from 127.0.0.1:40600

    15-Apr-2013 09:57:19.596 [CM503007]: Call(C:43): Ivr:1851 has joined, contact <sip:1851@127.0.0.1:40600>

    15-Apr-2013 09:57:19.594 [CM503007]: Call(C:43): Line:10002<<xxxxxxxxxx has joined, contact <sip:10002@192.168.225.5:5060>

    15-Apr-2013 09:57:19.592 L:43.2[Ivr] has joined to L:43.1[Line:10002<<xxxxxxxxxx]

    15-Apr-2013 09:57:19.591 NAT/ALG check:L:43.2[Ivr] RESPONSE 200 on 'INVITE' - basic check passed. No information for extended checks

    15-Apr-2013 09:57:19.447 [CM503025]: Call(C:43): Calling T:Ivr:1851@[Dev:sip:1851@127.0.0.1:40600;rinstance=2612db86cc34a073] for L:43.1[Line:10002<<xxxxxxxxxx]

    15-Apr-2013 09:57:19.400 [CM503027]: Call(C:43): From: Line:10002<<xxxxxxxxxx (<sip:xxxxxxxxxx@192.168.1.32:5060>) to T:Ivr:1851@[Dev:sip:1851@127.0.0.1:40600;rinstance=2612db86cc34a073]

    15-Apr-2013 09:57:19.400 [CM503004]: Call(C:43): Route 1: from L:43.1[Line:10002<<xxxxxxxxxx] to T:Ivr:1851@[Dev:sip:1851@127.0.0.1:40600;rinstance=2612db86cc34a073]

    15-Apr-2013 09:57:19.397 [CM503001]: Call(C:43): Incoming call from Line:10002<<xxxxxxxxxx to <sip:1851@192.168.225.2:5060>

    15-Apr-2013 09:57:19.395 Line limit check: Current # of calls for line Lc:10002(@DCT PRI 2[<sip:10002@192.168.225.5:5060>]) is 1; limit is 23

    15-Apr-2013 09:57:19.395 NAT/ALG check:L:43.1[Line:10002<<xxxxxxxxxx] REQUEST 'INVITE' - basic check passed. No information for extended checks

    15-Apr-2013 09:57:19.389 [CM503012]: Inbound any hours rule (unnamed) for 10002 forwards to DN:1851
     
  2. VAD_Support

    VAD_Support Active Member

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    In order to understand why a VAD application is failing, you need to look to the IVR log files first. The logs you're viewing are for the PBX, and only show that the call has been disconnected. But we need to understand why the script has ended prematurely.

    The following article explains how to troubleshoot problems with VAD applications:
    http://www.3cx.com/blog/voip-howto/troubleshooting-vad-applications/

    Please follow the steps in that article. If you still can't find the issue let us know and we'll investigate further.

    Thanks,
     
  3. EdBirchall

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    Thankyou for your reply. Looking at the logs, it appears our issue stems from a call to a SQL database that we are trying to make.
    Using the SESSION.ANI, we match the area code & exchange to anentry within the database. This then gives us an external number to which the call should be transferred.

    We have a database access component within which we set up the SQL query but we are getting the following errors:
    10:17:37.017|5864|(0):Error! Module '3CX.com.OSBjsi'. Error ID 501. errmsgmissing ; before statementline1linetxt192.168.1.50tokentxt.1.50

    10:17:37.017|5864|(0):Error! Module '3CX.com.OSBjsi'. Error ID 501. errmsgSyntaxError: missing ; before statementline1linetxt192.168.1.50tokentxt.1.50

    We believe we've set everything up correctly, using the manual and are once again uncertain as to what is going wong. Once again, help is greatly appreciated.
     
  4. VAD_Support

    VAD_Support Active Member

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    It seems that you're entering an IP address without quotes. For example:
    192.168.1.50
    instead of
    '192.168.1.50'

    Please note that the server field is an expression, so you need to enter a valid javascript expression there. If you want to use a constant string literal, then use single quotes arround it.

    Regards,
     
  5. EdBirchall

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    We put in the quotes which cleared up the error. Now, the call we are enabling just ends without any action being taken. We looked at the log file and it shows only the following messages:

    15:22:08.152|2264|PbxMedia\IVREngine.cpp(1609):IvrService, IVREngine::RegisterLines(), Registering of lines...127.0.0.1:5060127.0.0.1:50609999[HOL,7777,851,852,1852,1851,1853][][EndCall,IVRForward,MakeCall,PlayFile,RecordFile]

    15:22:08.152|2264|PbxMedia\IVREngine.cpp(1735):IvrService, IVREngine::RegisterLines(), Registering of lines done.

    We have the verbose setting set within the .ini file.

    We then tried to build in debug mode but, after a succeeful build, we get the following pop up screen. Reading other logs, this suggests a possible firewall issue but we have been assured that all ports are open.

    An error occurred trying to get the Debug Information Files from server:
    Inable to connect to the remote server. No connection could be made beacuse the target machine actively refused it. 127.0.0.1:5000
     
  6. VAD_Support

    VAD_Support Active Member

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    The logs attached are not in verbose mode. Have you restarted the 3CX services after changing the log configuration?
     
  7. EdBirchall

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    Thanks for your help this far. We now have a situation where we transfer a call to the phone number of a specific sales rep based on data from the database. However, if that line is busy, we need to prompt the caller either to leave a voice mail message for the rep or transfer to customer service.

    It seems that a busy signal invokes the ErrorHandler but how do we use the ErrorHandler to perform the above scenario?
     
  8. VAD_Support

    VAD_Support Active Member

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    The VAD only performs blind transfers. If the destination is busy, the VAD will not know it. The flow will continue in the disconnect handler flow, in case you need to perform some non-telephony task, like writing something to a database for example.

    In this case what you could do is configure the agent extension to do something different when busy. For example, you can configure it to transfer the call to another VAD callflow, where you can ask the user if he/she wants to transfer the call to customer service or leave a voicemail.

    Regards,
     
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