Conference number behind sip trunk doesn't works

Discussion in 'General' started by apollo, Mar 24, 2010.

  1. apollo

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    Hi,

    I would like to use elastix in order to make a conference server connected to an aastra pabx.
    I created a sip trunk connected to the aastra pabx (For the aastra, elastix is like a SIP phone)

    I created an inbound route that route all the calls (any CID / any DID) to the conference number.
    When I try to call the sip trunk number from the aastra pabx, the line is busy.

    If I change the inbound route in order to route all the calls to an elastix extension, it works. (I can make a conversation between a phone connected to the aastra and the phone connected to the elastix)

    If I call the conference number from the phone connected to the elastix, it works.

    I don't understand why I can call an elastix phone extension from the aastra and why I can't call the conference extension.

    I enabled sip debug and I can see that the aastra send an BYE request after it sends a ACK request.

    Thanks
     
  2. rollinsolo

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    If I were you I would post this topic/question in the Hardware/Endpoints/aastra section, because this mostly pertains to the Aastra PBX not elastix, Most here only use Elastix for most functionality and Aastra IP phones to connect to the box. Try there should be more support for you.
     
  3. apollo

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    I moved the question to the aastra forum.
     
  4. dicko

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    It's a matter of contexts and inclusions, ext-meetme is an internal context by default you would need to add it to from-pstn-custom to allow an inbound trunk access.
     

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