== Using SIP RTP CoS mark 5
    -- Executing [2302@milktest:1] Dial("SIP/2303-00000013", "SIP/2302,60,Tt") in new stack
  == Using SIP RTP CoS mark 5
    -- Called 2302
    -- SIP/2302-00000014 is ringing
callmanager*CLI>
callmanager*CLI>
callmanager*CLI>
    -- SIP/2302-00000014 answered SIP/2303-00000013
    -- Remotely bridging SIP/2303-00000013 and SIP/2302-00000014
callmanager*CLI>
callmanager*CLI>
callmanager*CLI>
    -- Remotely bridging SIP/2303-00000013 and SIP/2302-00000014
    -- Remotely bridging SIP/2303-00000013 and SIP/2302-00000014
    -- Started music on hold, class 'default', on SIP/2303-00000013
    -- <SIP/2302-00000014> Playing 'pbx-transfer.ulaw' (language 'en')
    -- Executing [2301@milktest:1] Dial("Local/2301@milktest-9d49;2", "SIP/2301,60,Tt") in new stack
  == Using SIP RTP CoS mark 5
    -- Called 2301
    -- SIP/2301-00000015 is ringing
    -- Local/2301@milktest-9d49;1 is ringing
callmanager*CLI>
callmanager*CLI>
callmanager*CLI>
callmanager*CLI>
callmanager*CLI>
callmanager*CLI>
callmanager*CLI>
    -- Got SIP response 480 "Temporarily Unavailable" back from 172.23.49.101:16398
    -- SIP/2301-00000015 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [2301@milktest:2] Hangup("Local/2301@milktest-9d49;2", "") in new stack
  == Spawn extension (milktest, 2301, 2) exited non-zero on 'Local/2301@milktest-9d49;2'
[Nov 29 18:39:32] NOTICE[5499]: features.c:2114 builtin_atxfer: We're trying to call SIP/2302
  == Using SIP RTP CoS mark 5
[Nov 29 18:39:32] NOTICE[5499]: chan_sip.c:5513 update_call_counter: Call to peer '2302' rejected due to usage limit of 1
[Nov 29 18:39:32] NOTICE[5499]: features.c:2793 feature_request_and_dial: Unable to call channel SIP/2302
    -- Stopped music on hold on SIP/2303-00000013
    -- Started music on hold, class 'default', on SIP/2303-00000013
    -- Executing [2301@milktest:1] Dial("Local/2301@milktest-343e;2", "SIP/2301,60,Tt") in new stack
  == Using SIP RTP CoS mark 5
    -- Called 2301
    -- SIP/2301-00000017 is ringing
    -- Local/2301@milktest-343e;1 is ringing
callmanager*CLI>
callmanager*CLI>
callmanager*CLI>
callmanager*CLI>
callmanager*CLI>
callmanager*CLI>
callmanager*CLI>
callmanager*CLI>
callmanager*CLI>
    -- Got SIP response 480 "Temporarily Unavailable" back from 172.23.49.101:16398
    -- SIP/2301-00000017 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [2301@milktest:2] Hangup("Local/2301@milktest-343e;2", "") in new stack
  == Spawn extension (milktest, 2301, 2) exited non-zero on 'Local/2301@milktest-343e;2'
callmanager*CLI>
callmanager*CLI>
callmanager*CLI>
callmanager*CLI>
callmanager*CLI>
callmanager*CLI>
callmanager*CLI>
  == Using SIP RTP CoS mark 5
[Nov 29 18:39:47] NOTICE[5499]: chan_sip.c:5513 update_call_counter: Call to peer '2302' rejected due to usage limit of 1
[Nov 29 18:39:47] NOTICE[5499]: features.c:2793 feature_request_and_dial: Unable to call channel SIP/2302
    -- Stopped music on hold on SIP/2303-00000013
    -- Started music on hold, class 'default', on SIP/2303-00000013
    -- Executing [2301@milktest:1] Dial("Local/2301@milktest-de8b;2", "SIP/2301,60,Tt") in new stack
  == Using SIP RTP CoS mark 5
    -- Called 2301
    -- SIP/2301-00000019 is ringing
    -- Local/2301@milktest-de8b;1 is ringing
    -- Got SIP response 480 "Temporarily Unavailable" back from 172.23.49.101:16398
    -- SIP/2301-00000019 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [2301@milktest:2] Hangup("Local/2301@milktest-de8b;2", "") in new stack
  == Spawn extension (milktest, 2301, 2) exited non-zero on 'Local/2301@milktest-de8b;2'
callmanager*CLI>
callmanager*CLI>
callmanager*CLI>
callmanager*CLI>
[Nov 29 18:39:52] NOTICE[5324]: chan_sip.c:23270 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 2301
  == Using SIP RTP CoS mark 5
[Nov 29 18:40:00] NOTICE[5499]: chan_sip.c:5513 update_call_counter: Call to peer '2302' rejected due to usage limit of 1
[Nov 29 18:40:00] NOTICE[5499]: features.c:2793 feature_request_and_dial: Unable to call channel SIP/2302
    -- Stopped music on hold on SIP/2303-00000013
[Nov 29 18:40:00] NOTICE[5499]: features.c:3520 ast_bridge_call: roger: call ast_cdr_end at 2 before h=0
  == Spawn extension (milktest, 2302, 1) exited non-zero on 'SIP/2303-00000013'