Trunks are not working asterisk SIP

Hello everyone,
we have a panasonic IP pbx KX-TDE600 one asterisk sip, and three more IP pbx in direrent location. all the trunks were working fine. suddenly from last weekly we are not able to call any extension from panasonic pbx to SIP trunks but sip trunks extension can call panasonic pbx but panasonic pbx can call sip extension. whenever we are calling to SIP trunk logs are generating in asterisk server it means calls are forwarded to SIP server from panasonic ip pbx. but in SIP server it is saying “TRUNK Dial failed due to CONGESTION - failing through to other trunks”. there is not issue with connectivity and no new changes till now. I am sharing complete log of a failed call of panasonic to SIP trunks. seeking help from you guyz. Thanks in advance.

-- Executing [[email protected]:1] Set("SIP/PABX-00000038", "INTRACOMPANYROUTE=YES") in new stack
-- Executing [[email protected]:2] Macro("SIP/PABX-00000038", "user-callerid|SKIPTTL|") in new stack
-- Executing [[email protected]:1] Set("SIP/PABX-00000038", "AMPUSER=pabx") in new stack
-- Executing [[email protected]:2] GotoIf("SIP/PABX-00000038", "0?report") in new stack
-- Executing [[email protected]:3] ExecIf("SIP/PABX-00000038", "1|Set|REALCALLERIDNUM=pabx") in new stack
-- Executing [[email protected]:4] Set("SIP/PABX-00000038", "AMPUSER=") in new stack
-- Executing [[email protected]:5] Set("SIP/PABX-00000038", "AMPUSERCIDNAME=") in new stack
-- Executing [[email protected]:6] GotoIf("SIP/PABX-00000038", "1?report") in new stack
-- Goto (macro-user-callerid,s,10)
-- Executing [[email protected]:10] GotoIf("SIP/PABX-00000038", "1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [[email protected]:19] NoOp("SIP/PABX-00000038", "Using CallerID "Habib-IT" <pabx>") in new stack
-- Executing [[email protected]:3] Set("SIP/PABX-00000038", "_NODEST=") in new stack
-- Executing [[email protected]:4] Macro("SIP/PABX-00000038", "record-enable||OUT|") in new stack
-- Executing [[email protected]:1] GotoIf("SIP/PABX-00000038", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [[email protected]:4] AGI("SIP/PABX-00000038", "recordingcheck|20170518-140253|1495094573.56") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20170518-140253|1495094573.56: No AMPUSER db entry for . Not recording
-- AGI Script recordingcheck completed, returning 0
-- Executing [[email protected]:5] MacroExit("SIP/PABX-00000038", "") in new stack
-- Executing [[email protected]:5] Macro("SIP/PABX-00000038", "dialout-trunk|2|2210||") in new stack
-- Executing [[email protected]:1] Set("SIP/PABX-00000038", "DIAL_TRUNK=2") in new stack
-- Executing [[email protected]:2] GosubIf("SIP/PABX-00000038", "0?sub-pincheck|s|1") in new stack
-- Executing [[email protected]:3] GotoIf("SIP/PABX-00000038", "0?disabletrunk|1") in new stack
-- Executing [[email protected]:4] Set("SIP/PABX-00000038", "DIAL_NUMBER=2210") in new stack
-- Executing [[email protected]:5] Set("SIP/PABX-00000038", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing [[email protected]:6] Set("SIP/PABX-00000038", "OUTBOUND_GROUP=OUT_2") in new stack
-- Executing [[email protected]:7] GotoIf("SIP/PABX-00000038", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [[email protected]:9] GotoIf("SIP/PABX-00000038", "1?skipoutcid") in new stack
-- Goto (macro-dialout-trunk,s,12)
-- Executing [[email protected]:12] ExecIf("SIP/PABX-00000038", "1|AGI|fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
-- AGI Script fixlocalprefix completed, returning 0
-- Executing [[email protected]:13] Set("SIP/PABX-00000038", "OUTNUM=2210") in new stack
-- Executing [[email protected]:14] Set("SIP/PABX-00000038", "custom=SIP/KHULNA") in new stack
-- Executing [[email protected]:15] ExecIf("SIP/PABX-00000038", "0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)tr") in new stack
-- Executing [[email protected]:16] Macro("SIP/PABX-00000038", "dialout-trunk-predial-hook|") in new stack
-- Executing [[email protected]:1] MacroExit("SIP/PABX-00000038", "") in new stack
-- Executing [[email protected]:17] GotoIf("SIP/PABX-00000038", "0?bypass|1") in new stack
-- Executing [[email protected]:18] GotoIf("SIP/PABX-00000038", "0?customtrunk") in new stack
-- Executing [[email protected]:19] Dial("SIP/PABX-00000038", "SIP/KHULNA/2210|300|tr") in new stack
-- Called KHULNA/2210
-- Got SIP response 503 "Service Unavailable" back from 172.29.11.32
-- SIP/KHULNA-00000039 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [[email protected]:20] Goto("SIP/PABX-00000038", "s-CONGESTION|1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing [[email protected]:1] GotoIf("SIP/PABX-00000038", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,3)
-- Executing [[email protected]:3] NoOp("SIP/PABX-00000038", "TRUNK Dial failed due to CONGESTION - failing through to other trunks") in new stack
-- Executing [[email protected]:6] Macro("SIP/PABX-00000038", "outisbusy|") in new stack
-- Executing [[email protected]:1] Playback("SIP/PABX-00000038", "all-circuits-busy-now|noanswer") in new stack
-- <SIP/PABX-00000038> Playing 'all-circuits-busy-now' (language 'en')
-- Executing [[email protected]:2] Playback("SIP/PABX-00000038", "pls-try-call-later|noanswer") in new stack
-- <SIP/PABX-00000038> Playing 'pls-try-call-later' (language 'en')
== Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/PABX-00000038' in macro 'outisbusy'
== Spawn extension (from-internal, 2210, 6) exited non-zero on 'SIP/PABX-00000038'
-- Executing [[email protected]:1] Macro("SIP/PABX-00000038", "hangupcall") in new stack
-- Executing [[email protected]:1] GotoIf("SIP/PABX-00000038", "1?endmixmoncheck") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [[email protected]:9] NoOp("SIP/PABX-00000038", "End of MIXMON check") in new stack
-- Executing [[email protected]:10] GotoIf("SIP/PABX-00000038", "1?nomeetmemon") in new stack
-- Goto (macro-hangupcall,s,12)
-- Executing [[email protected]:12] NoOp("SIP/PABX-00000038", "MEETME_RECORDINGFILE=") in new stack
-- Executing [[email protected]:13] GotoIf("SIP/PABX-00000038", "1?noautomon") in new stack
-- Goto (macro-hangupcall,s,15)
-- Executing [[email protected]:15] NoOp("SIP/PABX-00000038", "TOUCH_MONITOR_OUTPUT=") in new stack
-- Executing [[email protected]:16] GotoIf("SIP/PABX-00000038", "1?noautomon2") in new stack
-- Goto (macro-hangupcall,s,18)
-- Executing [[email protected]:18] NoOp("SIP/PABX-00000038", "MONITOR_FILENAME=") in new stack
-- Executing [[email protected]:19] GotoIf("SIP/PABX-00000038", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,22)
-- Executing [[email protected]:22] GotoIf("SIP/PABX-00000038", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,25)
-- Executing [[email protected]:25] GotoIf("SIP/PABX-00000038", "1?theend") in new stack
-- Goto (macro-hangupcall,s,27)
-- Executing [[email protected]:27] Hangup("SIP/PABX-00000038", "") in new stack
== Spawn extension (macro-hangupcall, s, 27) exited non-zero on 'SIP/PABX-00000038' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/PABX-00000038'

Check your trunks to see if they are connecting correctly. It looks like a password (or something) has changed and the connection is no longer getting made. I made one of these last night on an up-to-date server using a Chan-SIP connection, so I’m pretty sure it isn’t an issue with FreePBX.

There have been many reports recently about issues with PJ-SIP and trunks. Well, not many, but enough to erode what little confidence I had in PJ-SIP, but either way check your trunk configuration and make sure you’re connecting via Chan-SIP and not PJ-SIP.

Hi, we have three similar SIP in remote location and facing same issue instantly for every sip trunk. So it should not be an issue remote sip. something wrong with my asterisk.all of my remote sip are connected with asterisk. Please check image.