SIP trunk , incoming works but not outgoing

Discussion in 'General' started by westcomlimited, Feb 18, 2011.

  1. westcomlimited

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    I am newish to elastix but have the task of testing a dedicated voice vlan and SIP trunk.

    I have been provided with a limited amount of information from the VSP but I have been able to get things working to a point.

    I am running the latest version of elastix at the time of writing this post (stable repo not beta)

    My Elastix server has 2 Nics in it. It was installed with one NIC and then I installed a second.
    eth0 is connected to my local network and the gateway is set to my local firewall. Elastix has internet access through eth0
    eth1 is connected to a VSP supplied cisco layer 3 switch. A dedicated Vlan is setup between this cisco switch and the VSP through our fibre optic cable. There is no internet access on eth1.

    eth1 IP = 10.12.176.10
    I have setup the eth0 IP and gateway according to the VSP's instructions.
    Their SIP media is on 10.20.61.1 behind a gateway of 10.12.176.252
    I have a static route that routes like this:
    10.20.61.1 gw 10.12.176.252 eth1

    I can ping 10.20.61.1

    I have setup incoming DDI's to some test extensions and can successfully call them from outside the phone network.
    I have experimented with the dial plans quite a lot and I think I have it sending the correct E.164 format to the VSP (see below)

    This is the error I get when calling out on the test system.(phone numbers changed naturally)
    == Using SIP RTP TOS bits 184
    == Using SIP RTP CoS mark 5
    -- Called myVSP/6441111111
    -- Got SIP response 603 "Decline" back from 10.20.61.1
    -- SIP/myVSP-0000000e is busy
    == Everyone is busy/congested at this time (1:1/0/0)

    There are two things I suspect now after all my troubleshooting:
    sip_nat.conf <======== what would I want to put in here given my current network config.
    The VSP supplied layer 3 switch could be blocking required incoming ports?
     
  2. dicko

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    Perhaps describe the sub-netting a little better, if your network is 10.0.0.0/8 then no gateway should be necessary to the media server.

    But a 603 is a response to a NOTIFY packet, in your case not accepted, you need to further expound on your "SIP media server" at 10.20.61.1, my guess is that it is mis-configured.
     
  3. westcomlimited

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    Sorry I forgot to put the gateway, was a lot to remember.

    255.255.255.0 is the subnet on eth1

    The SIP media device at 10.20.61.1 belongs to the VSP, I know nothing at all about it other than its IP address, that Registration is disabled and what codec it uses.

    The VSP couldn't even provide me Peer Details I had to get that information off someone else who had setup a trixbox connected to them.
     
  4. dicko

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    Given that you ave no information about anything useful on the sip proxy/gateway, and you have not posted any of your configurations, then all I can suggest is turning SIP debugging on:

    sip set debug ip 10.20.61.1

    and getting out the FM's. as you

    tail -f /var/log/asterisk/full

    sorry, dicko
     
  5. westcomlimited

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    Here is the Peer Details from the Trunk
    Outgoing=
    Dial Plan = 64+.
    type=friend
    insecure=very
    host=10.20.61.1
    dtmfmode=rfc2833
    canreinvite=no
    disallow=all
    allow=ulaw,alaw
    context=from-trunk

    Incoming
    user settings = blank

    Outgoing routes =
    3+NXXXXXX
    0|NXXXXXXX


    attached is a clipped version of the full.log http://forum.elastix.org/old_files/call_trace_short.txt
     
  6. dicko

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    I'm sory my friend you were rejected by the agent you are using , i.e.

    User-Agent: Callplus SBC


    with cause 21

    Cause No. 21 - call rejected.
    This cause indicates that the equipment sending this cause does not wish to accept this call. although it could have accepted the call because the equipment sending this cause is neither busy nor incompatible. This cause may also be generated by the network, indicating that the call was cleared due to a supplementary service constraint. The diagnostic field may contain additional information about the supplementary service and reason for rejection.

    There is very likely no possibility to diagnose this further here in an Asterisk forum. Please take it to whoever the session border controller that identifies itself as "Callplus SBC" is, good luck

    dicko
     
  7. westcomlimited

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    Thank you again Dicko. This is all good amunition for when I take this up with the Vsp
     

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