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lejab

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Posts posted by lejab

  1. I don't have a BT200; I think if you are on same LAN try not to use any NAT servers or set it to "No"

    see if that will help.

    That was how i first tried.

    I have now tried another PBX software, pbxnsip and attended transfer works great. I think the transfers are handled by the PBX in this software.

    But I still want to get it to work with the support of the BT200 and Axon.

    sipnpbx is way to expensive...

     

     

    //Micke

  2. Hi -

     

    I am using version 1.1.1.14 - attended transfers work fine with me - both internal only - or including a 'sip' external call

     

    Sip port : 5060

     

    Rip port : random

     

    Stun is on : stun.gossiptel.com

     

    Hope this helps

     

    if not please send me a pm and can discuss further

     

    ---------------------------------------------------------------------------

     

    also tested with Express Talk as an extension --- all fine

     

    with a 'normal' phone on a Netgear TA612 -- I get one way audio - or if the normal phone origianted call - everything disconnected

     

    I tried with random RTP port but no difference. I had the 1.1.1.14 firmware from the beginning but Grandstream sad there was a bug with attended transfer so I got the 1.1.2.23 firmware from them.

    Strange thing is that unattended transfer works fine.

    When using a analog phone connected to a Linksys RT31P2 as caller it also works fine so the problem must be with the BT200 and Axon.

    //Micke

  3. Hi,

    Is there anybody who can help me with this.

    We are using 5 BT200 f/w: 1.1.2.23 and Axon PBX v1.09.

    The problem is when we try to make an attended transfer we get one way audio.

     

    Scenario 1:

    A calls B and asks for C.

    B press "flash" and calls C.

    C answers and B is telling him that A wants to talk with him.

    B press "transfer" and hangup.

    A can now hear C, but C doesn´t hear A.

     

    All parts using BT200 on the same LAN as the server running Axon.

    No firewalls enabled.

    SIP port: 5060

    RTP port: 16384

     

    Scenario 2:

    A calls B and asks for C.

    B press "flash" and calls C.

    C answers and B is telling him that A wants to talk with him.

    B press "transfer" and hangup.

    A get´s disconnected and C get´s busy signal.

    The only difference is that A is an call from an external line.

     

    Normal transfer works fine but not attended.

     

    We have talked with Grandstream and got firmware 1.1.2.23 that was supposed to fix this bug thay say.

    But he problem still remains.

     

    //Micke

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