Problems with transfer and transfer to Vmail

  1. 14 years ago

    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

  2. admin

    28 Feb 2010 Administrator

    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,

  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': '' }

  4. admin

    28 Feb 2010 Administrator

    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,

  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

  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 !!

  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

  8. admin

    2 Mar 2010 Administrator

    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?

  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

  10. admin

    3 Mar 2010 Administrator

    Hi,

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

  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?

  12. admin

    15 Mar 2010 Administrator

    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

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