One Extension Unable to Receive Calls

I recently switched the phones at a remote office from a leased line back to the main office to a new connection with more bandwidth. I was able to successfully move all phones with the exception of one Cisco 7940, which can call out but receives a fast busy signal when trying to call it. Every phone at the remote office is currently registered. I have checked the configs and this phone is set up exactly like every other phone at the office. Below is the output from Asterisk when dialing from one of the remote extensions to the other. I get a similar result when dialing from a phone at the main office or from a cell phone:

== Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 6 -- Executing [[email protected]:1] Macro("SIP/122-0001185b", "exten-vm,novm,120") in new stack -- Executing [[email protected]:1] Macro("SIP/122-0001185b", "user-callerid") in new stack -- Executing [[email protected]:1] Set("SIP/122-0001185b", "AMPUSER=122") in new stack -- Executing [[email protected]:2] GotoIf("SIP/122-0001185b", "0?report") in new stack -- Executing [[email protected]:3] ExecIf("SIP/122-0001185b", "1?Set(REALCALLERIDNUM=122)") in new stack -- Executing [[email protected]:4] Set("SIP/122-0001185b", "AMPUSER=122") in new stack -- Executing [[email protected]:5] Set("SIP/122-0001185b", "AMPUSERCIDNAME=Conf Room") in new stack -- Executing [[email protected]:6] GotoIf("SIP/122-0001185b", "0?report") in new stack -- Executing [[email protected]:7] Set("SIP/122-0001185b", "AMPUSERCID=122") in new stack -- Executing [[email protected]:8] Set("SIP/122-0001185b", "CALLERID(all)="Conf Room" <122>") in new stack -- Executing [[email protected]:9] ExecIf("SIP/122-0001185b", "0?Set(CHANNEL(language)=)") in new stack -- Executing [[email protected]:10] GotoIf("SIP/122-0001185b", "0?continue") in new stack -- Executing [[email protected]:11] Set("SIP/122-0001185b", "__TTL=64") in new stack -- Executing [[email protected]:12] GotoIf("SIP/122-0001185b", "1?continue") in new stack -- Goto (macro-user-callerid,s,19) -- Executing [[email protected]:19] NoOp("SIP/122-0001185b", "Using CallerID "Conf Room" <122>") in new stack -- Executing [[email protected]:2] Set("SIP/122-0001185b", "RingGroupMethod=none") in new stack -- Executing [[email protected]:3] Set("SIP/122-0001185b", "VMBOX=novm") in new stack -- Executing [[email protected]:4] Set("SIP/122-0001185b", "EXTTOCALL=120") in new stack -- Executing [[email protected]:5] Set("SIP/122-0001185b", "CFUEXT=") in new stack -- Executing [[email protected]:6] Set("SIP/122-0001185b", "CFBEXT=") in new stack -- Executing [[email protected]:7] Set("SIP/122-0001185b", "RT=""") in new stack -- Executing [[email protected]:8] Macro("SIP/122-0001185b", "record-enable,120,IN") in new stack -- Executing [[email protected]:1] GotoIf("SIP/122-0001185b", "1?check") in new stack -- Goto (macro-record-enable,s,4) -- Executing [[email protected]:4] AGI("SIP/122-0001185b", "recordingcheck,20140415-144322,1397591002.79949") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck,20140415-144322,1397591002.79949: Inbound recording enabled. recordingcheck,20140415-144322,1397591002.79949: CALLFILENAME=20140415-144322-1397591002.79949 -- AGI Script recordingcheck completed, returning 0 -- Executing [[email protected]:999] MixMonitor("SIP/122-0001185b", "20140415-144322-1397591002.79949.wav,,") in new stack -- Executing [[email protected]:9] Macro("SIP/122-0001185b", "dial,"",tr,120") in new stack -- Executing [[email protected]:1] GotoIf("SIP/122-0001185b", "1?dial") in new stack -- Goto (macro-dial,s,3) -- Executing [[email protected]:3] AGI("SIP/122-0001185b", "dialparties.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi == Begin MixMonitor Recording SIP/122-0001185b dialparties.agi: Starting New Dialparties.agi dialparties.agi: Caller ID name is 'Conf Room' number is '122' dialparties.agi: Methodology of ring is 'none' -- dialparties.agi: Added extension 120 to extension map -- dialparties.agi: Extension 120 cf is disabled -- dialparties.agi: Extension 120 do not disturb is disabled dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE) -- dialparties.agi: dbset CALLTRACE/120 to 122 -- dialparties.agi: Filtered ARG3: 120 -- AGI Script dialparties.agi completed, returning 0 -- Executing [[email protected]:7] Dial("SIP/122-0001185b", "SIP/120,"",tr") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 6 -- Called 120 -- SIP/120-0001185c is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [[email protected]:8] Set("SIP/122-0001185b", "DIALSTATUS=CONGESTION") in new stack -- Executing [[email protected]:9] GosubIf("SIP/122-0001185b", "0?CONGESTION,1") in new stack -- Executing [[email protected]:10] GotoIf("SIP/122-0001185b", "0?exit,return") in new stack -- Executing [[email protected]:11] Set("SIP/122-0001185b", "SV_DIALSTATUS=CONGESTION") in new stack -- Executing [[email protected]:12] GosubIf("SIP/122-0001185b", "0?docfu,1") in new stack -- Executing [[email protected]:13] GosubIf("SIP/122-0001185b", "0?docfb,1") in new stack -- Executing [[email protected]:14] Set("SIP/122-0001185b", "DIALSTATUS=CONGESTION") in new stack -- Executing [[email protected]:15] NoOp("SIP/122-0001185b", "Voicemail is 'novm'") in new stack -- Executing [[email protected]:16] GotoIf("SIP/122-0001185b", "1?s-CONGESTION,1") in new stack -- Goto (macro-exten-vm,s-CONGESTION,1) -- Executing [[email protected]:1] NoOp("SIP/122-0001185b", "IVR_RETVM: IVR_CONTEXT: ") in new stack -- Executing [[email protected]:2] GotoIf("SIP/122-0001185b", "0?exit,1") in new stack -- Executing [[email protected]:3] PlayTones("SIP/122-0001185b", "congestion") in new stack -- Executing [[email protected]:4] Congestion("SIP/122-0001185b", "10") in new stack == Spawn extension (macro-exten-vm, s-CONGESTION, 4) exited non-zero on 'SIP/122-0001185b' in macro 'exten-vm' == Spawn extension (from-internal, 120, 1) exited non-zero on 'SIP/122-0001185b' -- Executing [[email protected]:1] Macro("SIP/122-0001185b", "hangupcall") in new stack -- Executing [[email protected]:1] GotoIf("SIP/122-0001185b", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,4) -- Executing [[email protected]:4] GotoIf("SIP/122-0001185b", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,7) -- Executing [[email protected]:7] GotoIf("SIP/122-0001185b", "1?theend") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [[email protected]:9] Hangup("SIP/122-0001185b", "") in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/122-0001185b' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/122-0001185b' == MixMonitor close filestream == End MixMonitor Recording SIP/122-0001185b

It says the phone is busy. Perhaps DND is on.