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fangeli

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Posts posted by fangeli

  1. Thanks to all for the answers, now I understand where is the problem.

    I need to figure out how to use the "Send call to extension" option to route the incoming calls to various extensions.

    Currently if I call the extension 601, 602 or 603 from the main pbx, the call stops within the snom one with the error mentioned in the first post.

    I have read the document on the management of inbound call, but I have not figured out how to do this configuration.


    Thanks to all

  2. The problem is definitely an incorrect configuration.

    I created a sip trunk between an Asterisk and a snom one , the first have two trunks (one inbound and one outbound) type "SIP Gateway"

    pointing to a single trunk sull'asterisk .

    Calls from an extension registered on snom one, are rotated correctly toward the Asterisk, while the opposite does not happen.

    On the snom one I get the following error


    [8] 2013/12/20 12:33:26: Could not find a trunk (2 trunks)

    [7] 2013/12/20 12:33:26: Set packet length to 20

    [6] 2013/12/20 12:33:26: Call-leg 24: Sending RTP for 5d667fb07f40a34e5884ff734539d1d8@192.168.XXX.4:5060 to 192.168.XXX.4:13694, codec not set yet

    [8] 2013/12/20 12:33:26: Incoming call: Request URI sip:601@192.168.YYY.4, To is <sip:601@192.168.YYY.4>

    [5] 2013/12/20 12:33:26: Received incoming call without trunk information and user has not been found

    [8] 2013/12/20 12:33:26: call port 24: state code from 0 to 404

    [7] 2013/12/20 12:33:26: Set packet length to 20

    [8] 2013/12/20 12:33:26: Hangup: Call 24 not found

    [8] 2013/12/20 12:33:26: Clearing call port 24, SIP call id 5d667fb07f40a34e5884ff734539d1d8@192.168.XXX.4:5060


    I have checked all of the trunk and outbound route of the Asterisk, without finding obvious problems.


    Thank for the help


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