Conference number behind sip trunk doesn't works

Discussion in 'IP Phones' started by apollo, Mar 25, 2010.

  1. apollo

    Mar 24, 2010
    Likes Received:

    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.

  2. aastra1

    Mar 5, 2009
    Likes Received:
    Hello appolo,

    I believe this is a context issue as the meetme application (by default) only accept internal calls (from-internal) so when you call from the Aastra PBX you probably are presented as from-trunk so you get a busy signal.

    Check which context you are coming from and then modify the dialing plan to add ext-meetme. There are lots of explanations in various forums about that problem, just google 'asterisk meetme outside'. This is more a pure Asterisk question than an endpoint but we are here to help.

    My 2 cents.



Share This Page