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FOP2 Community Forum

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  3. Problems with transfer and transfer to Vmail

Problems with transfer and transfer to Vmail

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  • 6
    6
    6string
    wrote on last edited by
    #1

    I am using Elastix 1.5 Asterisk 1.4
    I have purchased a license
    All of the functions seem to work except the transfer and transfer to vmail
    I have tried the blind_transfer=1 in the fop2.cfg ( although I am on asterisk 1.4)
    I have tried the supervised_transfer=1
    I have upgraded to the beta version

    If I restart the fop2 service the transfer or transfer to vmail will work one time then nothing happens when using those 2 buttons. I don't see any response when I run a tail full -f

    Can anyone help
    Thanks for a really cool flash panel

    1 Reply Last reply
    0
    • N
      N
      nicolas
      wrote on last edited by
      #2

      Hi,

      If you use asterisk 1.4 unpatched, you must set supervised_transfer=0 and leave it that way. You have to start fop2_server in debug mode to see the manager events and potential problems:

      service fop2 stop
      /usr/local/fop2/fop2_server -X 15

      That will give us a hint on what the problem is. Best regards,

      1 Reply Last reply
      0
      • 6
        6
        6string
        wrote on last edited by
        #3

        Thanks for your response, here is what I have, Pressing transfer to voicemail
        I put the XXX in the remote address

        127.0.0.1 <- Event: PeerStatus
        127.0.0.1 <- Privilege: system,all
        127.0.0.1 <- Peer: SIP/10
        127.0.0.1 <- PeerStatus: Registered
        127.0.0.1 <- Server: 0

        127.0.0.1 <- Event: PeerStatus
        127.0.0.1 <- Privilege: system,all
        127.0.0.1 <- Peer: SIP/10
        127.0.0.1 <- PeerStatus: Registered
        127.0.0.1 <- Server: 0

        XXX.64.90.134 <= <msg data="7|tovoicemail|3|70de7017deaa2b01190cf4fcd1c733ce" />

        127.0.0.1 -> Action: Redirect
        127.0.0.1 -> Channel: Local/FMPR-18@from-internal-feb6
        127.0.0.1 -> Exten: *12
        127.0.0.1 -> Context: default
        127.0.0.1 -> Priority: 1
        127.0.0.1 -> Async: True

        127.0.0.1 <- Response: Error
        127.0.0.1 <- Message: Channel does not exist: Local/FMPR-18@from-internal-feb6
        127.0.0.1 <- Server: 0

        Response: Error
        Message: Channel does not exist: Local/FMPR-18@from-internal-feb6
        Server: 0
        XXX.64.90.134 <= <msg data="1|ping||" />
        XXX.64.90.134 => { 'btn': '0', 'cmd': 'pong', 'data': '0', 'slot': '' }

        127.0.0.1 <- Event: Registry
        127.0.0.1 <- Privilege: system,all
        127.0.0.1 <- ChannelDriver: SIP
        127.0.0.1 <- Domain: inbound17.vitelity.net
        127.0.0.1 <- Status: Registered
        127.0.0.1 <- Server: 0
        127.0.0.1 -> Action: Redirect
        127.0.0.1 -> Channel: Local/FMPR-18@from-internal-feb6
        127.0.0.1 -> Exten: *12
        127.0.0.1 -> Context: default
        127.0.0.1 -> Priority: 1
        127.0.0.1 -> Async: True

        127.0.0.1 <- Response: Error
        127.0.0.1 <- Message: Channel does not exist: Local/FMPR-18@from-internal-feb6
        127.0.0.1 <- Server: 0

        Response: Error
        Message: Channel does not exist: Local/FMPR-18@from-internal-feb6
        Server: 0

        XXX.64.90.134 <= <msg data="1|ping||" />
        XXX.64.90.134 => { 'btn': '0', 'cmd': 'pong', 'data': '0', 'slot': '' }

        1 Reply Last reply
        0
        • N
          N
          nicolas
          wrote on last edited by
          #4

          Ok. You are the 3rd person reporting this issue. However I was unable to reproduce the problem on my test system. I notice you are using followme, and that is a potential factor as it uses Local channels. However in my system I can transfer calls even if I have follow me enabled.

          I must be able to reproduce the issue in order to work on a fix. Can you tell me exactly what versions are you running of everything? FreePbx version, Asterisk version. And how is the call being made and your extensions setup, including followme?

          Do you call directly from one extension to the other? Do you call a did number, from there to an ivr and then your extension? Do you use ringgroups? Perhaps a combination of ring group and follow me?

          Best regards,

          1 Reply Last reply
          0
          • 6
            6
            6string
            wrote on last edited by
            #5

            OK, thanks for the response
            I am using Asterisk 1.4.24
            Freepbx 2.5.1.5
            Elastix 1.5.2-2.3
            The calls on the previous trace go like this
            A sip DID from Vitelity to an IVR
            I dialed ext 18 from the IVR then answered the phone on ext 18
            The call shows up with caller ID on the FOP2 panel
            I click on the destination Ext 12 then click the Transfer to VMail

            I have also tried ext to ext calls with the same result
            Also I am not using ring groups

            From reading your response I disabled the follow me settings on a few exts and tested
            That appears to be the problem, the transfer and transfer to vmail works once I disabled the follow me
            I will test further to see if I can find any other information

            Thanks

            1 Reply Last reply
            0
            • 6
              6
              6string
              wrote on last edited by
              #6

              Just a little more information I found
              The follow me on the destination exts makes no difference
              However if follow me is defined (and has an additional ext or number defined) on the ext that is using FOP2 then it will fail
              on my system when trying to transfer or transfer to vmail

              This will solve most of my problem,
              Thanks for all your help !!

              1 Reply Last reply
              0
              • 6
                6
                6string
                wrote on last edited by
                #7

                Sorry to be a pest
                I have the transfer and transfer to a station working
                But now I cannot transfer to any Queue or to Park
                It just hangs up
                Here is the trace of a Sip DID being transferred to Queue 50
                XXX.XXX.80.146 <= <msg data="1|atxfer|23|f3d868c14c77785f89883320076e82ae" />

                127.0.0.1 -> Action: Redirect
                127.0.0.1 -> Channel: SIP/dannylarsen-b7d27d80
                127.0.0.1 -> Exten: 50
                127.0.0.1 -> Context:
                127.0.0.1 -> Priority: 1
                127.0.0.1 -> Async: True

                127.0.0.1 <- Response: Success
                127.0.0.1 <- Message: Redirect successful
                127.0.0.1 <- Server: 0

                Response: Success
                Message: Redirect successful
                Server: 0

                127.0.0.1 <- Event: Unlink
                127.0.0.1 <- Privilege: call,all
                127.0.0.1 <- Channel1: SIP/dannylarsen-b7d27d80
                127.0.0.1 <- Channel2: SIP/10-b7b01d78
                127.0.0.1 <- Uniqueid1: 1267564614.458
                127.0.0.1 <- Uniqueid2: 1267564615.461
                127.0.0.1 <- CallerID1: 8012053201
                127.0.0.1 <- CallerID2: 10
                127.0.0.1 <- Server: 0

                127.0.0.1 <- Event: ExtensionStatus
                127.0.0.1 <- Privilege: call,all
                127.0.0.1 <- Exten: 10
                127.0.0.1 <- Context: ext-local
                127.0.0.1 <- Status: 0
                127.0.0.1 <- Server: 0

                127.0.0.1 <- Event: Hangup
                127.0.0.1 <- Privilege: call,all
                127.0.0.1 <- Channel: SIP/10-b7b01d78
                127.0.0.1 <- Uniqueid: 1267564615.461
                127.0.0.1 <- Cause: 16
                127.0.0.1 <- Cause-txt: Normal Clearing
                127.0.0.1 <- Server: 0

                127.0.0.1 <- Event: QueueMemberStatus
                127.0.0.1 <- Privilege: agent,all
                127.0.0.1 <- Queue: 50
                127.0.0.1 <- Location: SIP/10
                127.0.0.1 <- MemberName: SIP/10
                127.0.0.1 <- Membership: dynamic
                127.0.0.1 <- Penalty: 0
                127.0.0.1 <- CallsTaken: 3
                127.0.0.1 <- LastCall: 1267561947
                127.0.0.1 <- Status: 1
                127.0.0.1 <- Paused: 0
                127.0.0.1 <- Server: 0

                127.0.0.1 <- Event: Newexten
                127.0.0.1 <- Privilege: call,all
                127.0.0.1 <- Channel: SIP/dannylarsen-b7d27d80
                127.0.0.1 <- Context: macro-dial
                127.0.0.1 <- Extension: h
                127.0.0.1 <- Priority: 1
                127.0.0.1 <- Application: Macro
                127.0.0.1 <- AppData: hangupcall
                127.0.0.1 <- Uniqueid: 1267564614.458
                127.0.0.1 <- Server: 0

                127.0.0.1 <- Event: Newexten
                127.0.0.1 <- Privilege: call,all
                127.0.0.1 <- Channel: SIP/dannylarsen-b7d27d80
                127.0.0.1 <- Context: macro-hangupcall
                127.0.0.1 <- Extension: s
                127.0.0.1 <- Priority: 1
                127.0.0.1 <- Application: ResetCDR
                127.0.0.1 <- AppData: vw
                127.0.0.1 <- Uniqueid: 1267564614.458
                127.0.0.1 <- Server: 0

                127.0.0.1 <- Event: Newexten
                127.0.0.1 <- Privilege: call,all
                127.0.0.1 <- Channel: SIP/dannylarsen-b7d27d80
                127.0.0.1 <- Context: macro-hangupcall
                127.0.0.1 <- Extension: s
                127.0.0.1 <- Priority: 2
                127.0.0.1 <- Application: NoCDR
                127.0.0.1 <- AppData:
                127.0.0.1 <- Uniqueid: 1267564614.458
                127.0.0.1 <- Server: 0

                127.0.0.1 <- Event: Newexten
                127.0.0.1 <- Privilege: call,all
                127.0.0.1 <- Channel: SIP/dannylarsen-b7d27d80
                127.0.0.1 <- Context: macro-hangupcall
                127.0.0.1 <- Extension: s
                127.0.0.1 <- Priority: 3
                127.0.0.1 <- Application: GotoIf
                127.0.0.1 <- AppData: 1?skiprg
                127.0.0.1 <- Uniqueid: 1267564614.458
                127.0.0.1 <- Server: 0

                127.0.0.1 <- Event: Newexten
                127.0.0.1 <- Privilege: call,all
                127.0.0.1 <- Channel: SIP/dannylarsen-b7d27d80
                127.0.0.1 <- Context: macro-hangupcall
                127.0.0.1 <- Extension: s
                127.0.0.1 <- Priority: 6
                127.0.0.1 <- Application: GotoIf
                127.0.0.1 <- AppData: 0?skipblkvm
                127.0.0.1 <- Uniqueid: 1267564614.458
                127.0.0.1 <- Server: 0

                127.0.0.1 <- Event: Newexten
                127.0.0.1 <- Privilege: call,all
                127.0.0.1 <- Channel: SIP/dannylarsen-b7d27d80
                127.0.0.1 <- Context: macro-hangupcall
                127.0.0.1 <- Extension: s
                127.0.0.1 <- Priority: 7
                127.0.0.1 <- Application: NoOp
                127.0.0.1 <- AppData: Cleaning Up Block VM Flag: BLKVM/10/SIP/dannylarsen-b7d27d80
                127.0.0.1 <- Uniqueid: 1267564614.458
                127.0.0.1 <- Server: 0

                127.0.0.1 <- Event: Newexten
                127.0.0.1 <- Privilege: call,all
                127.0.0.1 <- Channel: SIP/dannylarsen-b7d27d80
                127.0.0.1 <- Context: macro-hangupcall
                127.0.0.1 <- Extension: s
                127.0.0.1 <- Priority: 8
                127.0.0.1 <- Application: DBdel
                127.0.0.1 <- AppData: BLKVM/10/SIP/dannylarsen-b7d27d80
                127.0.0.1 <- Uniqueid: 1267564614.458
                127.0.0.1 <- Server: 0

                127.0.0.1 <- Event: Newexten
                127.0.0.1 <- Privilege: call,all
                127.0.0.1 <- Channel: SIP/dannylarsen-b7d27d80
                127.0.0.1 <- Context: macro-hangupcall
                127.0.0.1 <- Extension: s
                127.0.0.1 <- Priority: 9
                127.0.0.1 <- Application: GotoIf
                127.0.0.1 <- AppData: 1?theend
                127.0.0.1 <- Uniqueid: 1267564614.458
                127.0.0.1 <- Server: 0

                127.0.0.1 <- Event: Newexten
                127.0.0.1 <- Privilege: call,all
                127.0.0.1 <- Channel: SIP/dannylarsen-b7d27d80
                127.0.0.1 <- Context: macro-hangupcall
                127.0.0.1 <- Extension: s
                127.0.0.1 <- Priority: 11
                127.0.0.1 <- Application: Hangup
                127.0.0.1 <- AppData:
                127.0.0.1 <- Uniqueid: 1267564614.458
                127.0.0.1 <- Server: 0

                127.0.0.1 <- Event: Hangup
                127.0.0.1 <- Privilege: call,all
                127.0.0.1 <- Channel: SIP/dannylarsen-b7d27d80
                127.0.0.1 <- Uniqueid: 1267564614.458
                127.0.0.1 <- Cause: 16
                127.0.0.1 <- Cause-txt: Normal Clearing
                127.0.0.1 <- Server: 0

                1 Reply Last reply
                0
                • N
                  N
                  nicolas
                  wrote on last edited by
                  #8

                  XXX.XXX.80.146 <= <msg data="1|atxfer|23|f3d868c14c77785f89883320076e82ae" />

                  127.0.0.1 -> Action: Redirect
                  127.0.0.1 -> Channel: SIP/dannylarsen-b7d27d80
                  127.0.0.1 -> Exten: 50
                  127.0.0.1 ->** Context:**
                  127.0.0.1 -> Priority: 1
                  127.0.0.1 -> Async: True

                  Context is missing. Do you use manual buttons configuration or automatic? How does your queue button config looks like?

                  1 Reply Last reply
                  0
                  • 6
                    6
                    6string
                    wrote on last edited by
                    #9

                    I am using the automatic config
                    Here is my Fop2.cfg
                    [general]
                    ; AMI definitions
                    manager_host=127.0.0.1
                    manager_port=5038
                    manager_user=fop2
                    manager_secret=fop2secret
                    ;event_mask=call,agent

                    ; Daemon definitios
                    ; listen_port = 4445
                    ;restrict_host = www.asternic.org
                    ;web_dir = /var/www/html/operator/fop2

                    ; Global Config
                    language = en
                    poll_interval = 86400
                    poll_voicemail = 1
                    ;monitor_ipaddress = 0

                    ; Force blind transfer on asterisk 1.6
                    ; blind_transfer = 1

                    ; Force supervised transfer on asterisk 1.4
                    ; requires the atxfer manager backport patch
                    ; supervised_transfer = 0

                    ; Force delimiter for asterisk applications
                    ; force_parameter_delimiter = ","

                    ; When adding or removing members to a queue, fop2 will default to
                    ; AddQueueMember/RemoveQueueMember commands. If you set use_agentlogin
                    ; to 1, together with the QueueChannel in a button definition set to
                    ; an Agent number it will use AgentCallbackLogin and Agentlogoff instead.
                    ;
                    use_agentlogin = 0

                    ; Filename to use when start monitoring, you can use ${UNIQUEID},
                    ; ${ORIG_EXTENSION}, ${DEST_EXTENSION}
                    ; and date formats %Y %m %d to construct the filename.

                    ; Master Password that overrides any individual one
                    master_key = 6string

                    ; Settings for modifying the recording filename
                    ; Available variables are:
                    ; ${UNIQUEID} = Unique Id of the call
                    ; ${TIMESTAMP} = Unix Timestamp when the recording was initiated
                    ; ${DEST_EXTENSION} = Target extenstion being monitored
                    ; ${ORIG_EXTENSION} = Extension/User that started the recording (not
                    ; the other leg)
                    ; Date variables:
                    ; %Y 4 digits year
                    ; %y 2 digits year
                    ; %m 2 digits month
                    ; %d 2 digits day
                    ; %h 2 digits hour
                    ; %i 2 digits minute
                    ; %s 2 digits seconds

                    monitor_filename=g${DEST_EXTENSION}-${UNIQUEID}
                    monitor_format=wav
                    monitor_mix=true

                    1 Reply Last reply
                    0
                    • N
                      N
                      nicolas
                      wrote on last edited by
                      #10

                      Hi,

                      Run the autoconfig-buttons-freepbx.sh script at the linux cli and look for the queue button configuration and paste it here..

                      1 Reply Last reply
                      0
                      • J
                        J
                        jmituzas
                        wrote on last edited by
                        #11

                        Hi,

                        If you use asterisk 1.4 unpatched, you must set supervised_transfer=0 and leave it that way.

                        Will that work for asterisk 1.6 as well ?

                        and place within fop2.cfg file, correct?

                        1 Reply Last reply
                        0
                        • N
                          N
                          nicolas
                          wrote on last edited by
                          #12

                          supervised_transfer=0 is for disabling the use of the atxfer manager command in Asterisk 1.4

                          blind_transfer=1 is for forcing standard redirects (and disabling atxfer) on asterisk 1.6.2

                          1 Reply Last reply
                          0

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