asterisk console

       > doing dnsmgr_lookup for 'sipnet.ru'
       > ast_get_srv: SRV lookup for '_sip._udp.sipnet.ru' mapped to host sipnet.ru, port 5060
  == Using SIP RTP CoS mark 5
    -- Executing [79031234567@DLPN_MainPlan:1] Macro("SIP/6001-0000000e", "trunkdial-failover-0.3,SIP/trunk_1/79031234567,,trunk_1,") in new stack
    -- Executing [[email protected]:1] GotoIf("SIP/6001-0000000e", "0?1-fmsetcid,1") in new stack
    -- Executing [[email protected]:2] GotoIf("SIP/6001-0000000e", "0?1-setgbobname,1") in new stack
    -- Executing [[email protected]:3] Set("SIP/6001-0000000e", "CALLERID(num)=") in new stack
    -- Executing [[email protected]:4] Set("SIP/6001-0000000e", "CALLERID(all)=") in new stack
    -- Executing [[email protected]:5] GotoIf("SIP/6001-0000000e", "0?1-dial,1") in new stack
    -- Executing [[email protected]:6] Set("SIP/6001-0000000e", "CALLERID(all)=") in new stack
    -- Executing [[email protected]:7] Set("SIP/6001-0000000e", "CALLERID(all)=") in new stack
    -- Executing [[email protected]:8] Goto("SIP/6001-0000000e", "1-dial,1") in new stack
    -- Goto (macro-trunkdial-failover-0.3,1-dial,1)
    -- Executing [[email protected]:1] Dial("SIP/6001-0000000e", "SIP/trunk_1/79031234567") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/trunk_1/79031234567
    -- Got SIP response 500 "account has been moved to a remote system" back from 212.53.40.40:5060
    -- SIP/trunk_1-0000000f is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [[email protected]:2] GotoIf("SIP/6001-0000000e", "0 > 0 ?1-CONGESTION,1:1-out,1") in new stack
    -- Goto (macro-trunkdial-failover-0.3,1-out,1)
    -- Executing [[email protected]:1] Hangup("SIP/6001-0000000e", "") in new stack
  == Spawn extension (macro-trunkdial-failover-0.3, 1-out, 1) exited non-zero on 'SIP/6001-0000000e' in macro 'trunkdial-failover-0.3'
  == Spawn extension (DLPN_MainPlan, 79031234567, 1) exited non-zero on 'SIP/6001-0000000e'