Attended Transfer problem

Discussion in 'General' started by possebon, Apr 3, 2009.

  1. possebon

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    Hi all,

    I've been trying to use the Attended Transfer on Elastix 1.3-2.

    I can't do it working, it does not matter what I configure on FreePBX or on directly on files (/etc/asterisk/features_general_custom.conf
    )

    my features_general_custom.conf file:

    blindxfer=*5
    atxfer=*4
    automon=*1
    disconnect=**
    featuredigittimeout = 1200;
    transferdigittimeout => 1200 ;


    Any clue?

    thanks in advance

    Fernando Possebon
     
  2. ramoncio

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    How are your phones connected?
    Are they in a sip gateway? A fxs port in a card?
    Which codecs and dtmf mode do you use to connect the gateway to asterisk?
    How do you get your inbound calls in? A SIP trunk? A PSTN trunk? A PRI or BRI?

    Usually this problems are dtmf related.
    Maybe your SIP gateway doesn´t pass them correctly, because it may use the * key for some other internal functions. In this case try to change features_general_custom.conf to avoid using keys that might cause conflicts, as # or *
    Other common cause is using inband dtmf mode in conjunction with high compression voice codecs, as gsm or g729. It is better to use g711 or slin for inband mode.
    If it is a sip trunk you'll probably need to use rfc2833.
     
  3. possebon

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    Hi ramoncio, thanks for your reply.

    In fact there is no gateway (while) or trunks. The test that I'm doing is Extension 1501 calls to extension 1503. The person on extension 1503 would like to transfer this call to the extension 1509.

    This does not work.

    I'm using X-Lite and QuteCom softphones.

    The codecs in use are ulaw.

    Thanks again for any clue.

    Best regards

    Fernando
     
  4. ramoncio

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    Have you tried any other feature codes starting with the * key?
    You should configure dtmf mode rfc2833 both in the X-Lite and in the Elastix web gui.
    Do you hear the "Transfer" message?
     

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