[Asterisk-Users] Re: sip to sip, not comunication

[Walter Willis]

log the call

-- Unregistered SIP '1234'
    -- Registered SIP '1234' at 201.230.97.110 port 9906 expires 3600
    -- Added extension '1234' priority 1 to sipregistrations
    -- Saved useragent "xlite release 3010n stamp 19039" for peer 1234
    -- Executing Dial("SIP/1234-cfb0", "SIP/123 at domain.com.pe|20|tr")
in new stack
    -- Called 123 at sorcier.com.pe
    -- Unregistered SIP '12345'
    -- Nobody picked up in 20000 ms
  == Auto fallthrough, channel 'SIP/1234-cfb0' status is 'NOANSWER'