Conference issue

Hi, i have an issue with conferences and supervised transfers.
I've a Freepbx 6 32bit with a lot of extensions. Each of this extensions is associated to a conference room (Every user has one own room). This is possible with the freepbx commercial module called Class Of Service.

Our customer want that only the room owner can transfer other users in conference room . (The conference is enabled only to the owner. If i do a call from another extension to conference room, the module filter my request )
This is possible only with supervised transer. Blind transfer is filtered by Class Of Service module.

When the owner (for example extension number 25) do supervised transfer to conference room, the extension 25 enter in the room, then hangup : the user moved is rightly in the conference.

My issue is:
In the conference room button i see only extension 25 . This happen for every user that i want in the conference room.
Finally, when the conference starts, i see many times extension 25.
Later, when the conference is terminated, fop2 can't remove many times extension 25. The conference remains hanging.

Is it a bug of fop2?
Is there another way to associate one extension to one conference room (without using pin)?

Trusting in your experience
Sorry for my english, thanks.

Comments

  • Hi,

    Supervised transfers into conferences is a mess in Asterisk itself. It is not a problem with FOP2, the problem becomes apparent/visible with FOP2. When doing at attendant transfer into the conference, a Local/xxx channel is used that is associated with the channel that initiates the transfer, that link is kept for as long as the call is up (making the transferer channel appear as busy). And the user in meetme is the transferer .

    So it is all messed up. And not because of FOP2, you can just not use FOP2 and that will happen and you can verify it via "core show channels concise" or "meetme list xxx".

    So, you should NOT use supervised transfers to send to conference rooms. If your chose of modules to do things forces you to use supervised transfers, then ask those module developers to fix them so you can use blind transfer instead of supervised ones mantaining all those restrictions you need.

    In FOP2 it is possible to create groups or specific permissions to restrict transfers in that way, but that is done outside of asterisk and works only for FOP2.

    I cannot speak about any commercial module on FreePBX, you should request/demand a fix or find a solution to perform that restriction you need using blind transfers instead.

    Best regards,

  • Hi, I did some tests and I still have doubts.

    I tried to simulate this situation: I'm extension 62 and i want to do a conference call ( room 5000) with extensions 60 and 74. When I add the users via supervised transer I've got the following visual issue: [attachments: fop2 - conference issue]

    When I run asterisk console, core show channels concise returns:

    SIP/60-00002b68!from-internal!STARTMEETME!4!Up!ConfBridge!5000,,,!60!!!3!168!(None)!1436188824.100716

    SIP/74-00002b65!from-internal!STARTMEETME!4!Up!ConfBridge!5000,,,!74!!!3!201!(None)!1436188787.100709


    I think that is a fop2 issue because if I try to restart the fop2 service, the visual list of the conference users is corrected.[attachments: last image]

    Trusting in your experience,
    Lorenzo.

  • Hi,

    My test was using meetme, not confbridge. They are different technologies and won't generate the same events or channels after attendant transfers. I will test with confbridge and let you know, however, the result is the same as seen with Meetme except that the local channels are not listed on the core show channels, but most probably the events are fired in AMI with them.

    The fact that they are cleared on reload it is not an indication of anything, as FOP2 does an internal map between a local channel and the real channel. It has to do that because Asterisk fails to keep track of it. That map is destroyed on reload/restart, so from your core show channels list it will display data correctly (note that with meetme, after reload problem is NOT cleared as local channels are still persistent on asterisk itself).

    If confbridge fire events and then fails to fire the proper hangup/destroy event for those channels, then it is still an asterisk problem. I will test and let you know.

    Best regards,
  • I've done a test with meet me conference and the result is the same.

    SIP/74-00002c74!from-internal!STARTMEETME!4!Up!MeetMe!5000,,!74!!!3!124!(None)!1436255243.103591

    This is what returns core show channels concise. The visual problem is the same.

    Thanks for your patience,
    Lorenzo.
  • How are you performing the transfer? Using FOP2 or native asterisk atxfer feature codes or with your SIP phone button? Asterisk behaves quite differently in either case.
  • From different types of ip phones. For example yealink t21 - t23 or grandstream gxp280.
  • Hi, the problem is still existing, can we check ?

    The transfer is perfomed by function keys of the IP Phone and not via feautures codes.
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