Transfer anywhere not working...

bahuntbahunt
in Support
I have the latest version with Centos using Asterisk. I am unable to transfer to another ext. or external number when I try the number dials and nothing happens. The dial feature doesn't work either any help would be greatly appreciated.


127.0.0.1 -> Action: Ping

127.0.0.1 <- Response: Success
127.0.0.1 <- Ping: Pong
127.0.0.1 <- Timestamp: 1387901217.085849

7***********:1203 <= <msg data="1_11|dragatxfer|12|9e2e95717bce335f679554518595fe70" />

boton por canal SIP/23******** esta blessed
127.0.0.1 -> Action: Atxfer
127.0.0.1 -> Channel: SIP/2*****3264-0000015f
127.0.0.1 -> Exten: 2345
127.0.0.1 -> Context: outgoing
127.0.0.1 -> Priority: 1
127.0.0.1 -> Async: True

127.0.0.1 <- Response: Success
127.0.0.1 <- Message: Atxfer successfully queued

127.0.0.1 <- Event: VarSet
127.0.0.1 <- Privilege: dialplan,all
127.0.0.1 <- Channel: SIP/2******3264-0000015f
127.0.0.1 <- Variable: TRANSFER_CONTEXT
127.0.0.1 <- Value: outgoing
127.0.0.1 <- Uniqueid: 13**********.352

Flash clients connected: 1
Client ***********:****, user: 2**********@GENERAL, type: websockets

127.0.0.1 <- Event: Registry
127.0.0.1 <- Privilege: system,all
127.0.0.1 <- ChannelType: SIP
127.0.0.1 <- Username: ****
127.0.0.1 <- Domain: *****
127.0.0.1

Comments

  • nicolasnicolas
    Does 2345@outgoing exists in your dialplan? It seems to me that you have an incorrect configuration in your button contexts, or similar to that.

    asterisk -rx "dialplan show 2345@outgoing"

    Best regards,
  • bahuntbahunt
    2345 does exist in my dial plan. The number is dialed when I try to transfer but nothing happens....
  • nicolasnicolas
    Then you are missing the tT options on your original dialplan dial command. Or you do not have the atxfer feature enabled in /etc/asterisk/features.conf
  • bahuntbahunt
    I added tT to the extension and made sure atxfer was enabled and still get the same response...


    ATXFER using native atxfer in AMI for SIP/2-00007

    127.0.0.1 -> Action: Atxfer
    127.0.0.1 -> Channel: SIP/2-00007
    127.0.0.1 -> Exten: 23
    127.0.0.1 -> Context: C21LD
    127.0.0.1 -> Priority: 1
    127.0.0.1 -> Async: True

    127.0.0.1 <- Response: Success
    127.0.0.1 <- Message: Atxfer successfully queued

    127.0.0.1 <- Event: VarSet
    127.0.0.1 <- Privilege: dialplan,all
    127.0.0.1 <- SequenceNumber: 2670
    127.0.0.1 <- File: pbx.c
    127.0.0.1 <- Line: 11404
    127.0.0.1 <- Func: pbx_builtin_setvar_helper
    127.0.0.1 <- Channel: SIP/2-0007
    127.0.0.1 <- Variable: TRANSFER_CONTEXT
    127.0.0.1 <- Value: C21LD
    127.0.0.1 <- Uniqueid: 14664.1130

    TIMER asterisk_ami_connect

    ** Asterisk Manager logged in localhost for 415 seconds
  • nicolasnicolas
    FOP2 is sending the correct command (it seems, as I do not know your dialplan). If the atxfer is not working, you will have to look at your asterisk config. Your problem it is *not* fop2 related.

    Best regards,
  • nicolasnicolas
    Confirmed via live help: The dialplan was NOT including the tT options in the Dial command, making the attendant transfer not to work.

    So, for other users that might have issues, please check and re check many times that you are using the correct dial options and that the atxfer feature is enabled also.

    This was NOT a fop2 issue, but an asterisk misconfiguration.
  • bahuntbahunt
    thanks for the great support!
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