If you have configured in Asterisk (or you fron-end FreePBX) sip trunk provider of VoIP, but outbound link is not working, and in output:

# asterisk -rx "sip show peers"

you see that your sip trunk UNREACHABLE in the “Status” field, check the following settings:

  • Disable qualify option for the corresponding peer:

    This will disable the periodic check availability of the peer. If you then outgoing calls via this trunk earned, it means that the provider is not response for checking packets (asterisk used packages OPTIONS with zero length). Possible reasons for this, see below

  • If asterisk (FreePBX) behind NAT (any type), check the settings in the instructions of the external IP:
    • in FreePBX get the desired options on the path Settings -> Asterisk SIP settings
    • or in sip.conf:
  • In severe cases, you can always thoroughly investigate include debug mode in the console asterisk (asterisk -r):
    pbx*CLI> sip set debug on
    SIP Debugging enabled
    pbx*CLI> core set debug 99
    Core debug was 0 and is now 99
    pbx*CLI> core set verbose 99
    Verbosity was 0 and is now 99

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One thought on “SIP trunk UNREACHABLE

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