Transfer help

Discussion in 'General' started by saugortgarcia, Jun 28, 2010.

  1. saugortgarcia

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    hi,
    I have an Elastix installed behind an PBX Alcatel OXE. Both PBX are conected through SIP trunk. Users from both side can talk each other perfectly. Also I have this:

    a. An user from oxe dial 6311, the oxe send the call to elastix. Elastix display a message then transfer the call back to OXE, ext 859. Until tranfer use/consume/take/occupy just one channel (SIP-REFER), so I got after tranfer the OXE reports all sip channels free.

    b. An user from oxe dial 6342, the oxe send the call to elastix. Elastix deliver the call to an internal ext 6342. Then the user transfer the call (transfer button on softphone SJphone) to exten 859. Until tranfer use/consume/take/occupy just two channel (NO SIP-REFER/RE-INVTE?), so I got after tranfer the OXE reports two sip channels busy.

    c. An user from oxe dial 6342, the oxe send the call to elastix. Elastix deliver the call to an internal ext 6342. Then the user transfer the call (transfer button on softphone SJphone) to exten 6311. Until tranfer use/consume/take/occupy just two channel (SIP-REFER/RE-INVTE?), the OXE reports all sip channels free.

    So my question is:
    How can I make all transfer work like case a? In this way, we could optimize sip channels.

    Asterisk logs show that for case b, asterisk use "macro-dialout-trunk".


    Thanks for your help and comments http://forum.elastix.org/old_files/logs.zip
     
  2. saugortgarcia

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    Have someone a clue about this?
     

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