Upgradt to Asterisk 1.4.27 broke MOH

I originally installes AsteriskNow/FreePBX version 2.5 with asterisk 1.4.26, on Centos 5.4. Yesturday I upgraded to FreePBX 2.6.0.0 and it fixed several issues. However I still had a problem with CDR and Call Monitor. Since it was a Distro I did not have any source files to add the add-ons. So I manually downloaded the tar ball for Asterisk 1.4.27, and performed the following steps:

make clean
./configure
make
make install
reboot

Great news it fixed the CDR and Call Monitor, and it seemed like everything else was working. Then I discovered that I only have MOH in Queues

localhost*CLI> moh show files
Class: default
File: /var/lib/asterisk/moh/macroform-cold_day
File: /var/lib/asterisk/moh/macroform-the_simplicity
File: /var/lib/asterisk/moh/reno_project-system
File: /var/lib/asterisk/moh/macroform-robot_dity
File: /var/lib/asterisk/moh/manolo_camp-morning_coffee

The above looks good to me but when I press the hold button on a phone I do not get any logging in the CLI and avter 30 seconds the call disconnects.

-- Executing [[email protected]:1] Macro("SIP/3303-00000008", "exten-vm|3301|3301") in new stack
-- Executing [[email protected]:1] Macro("SIP/3303-00000008", "user-callerid") in new stack
-- Executing [[email protected]:1] Set("SIP/3303-00000008", "AMPUSER=3303") in new stack
-- Executing [[email protected]:2] GotoIf("SIP/3303-00000008", "0?report") in new stack
-- Executing [[email protected]:3] ExecIf("SIP/3303-00000008", "1|Set|REALCALLERIDNUM=3303") in new stack
-- Executing [[email protected]:4] Set("SIP/3303-00000008", "AMPUSER=3303") in new stack
-- Executing [[email protected]:5] Set("SIP/3303-00000008", "AMPUSERCIDNAME=Admissions 3303") in new stack
-- Executing [[email protected]:6] GotoIf("SIP/3303-00000008", "0?report") in new stack
-- Executing [[email protected]:7] Set("SIP/3303-00000008", "AMPUSERCID=3303") in new stack
-- Executing [[email protected]:8] Set("SIP/3303-00000008", "CALLERID(all)="Admissions 3303" <3303>") in new stack
-- Executing [[email protected]:9] GotoIf("SIP/3303-00000008", "0?continue") in new stack
-- Executing [[email protected]:10] Set("SIP/3303-00000008", "__TTL=64") in new stack
-- Executing [[email protected]:11] GotoIf("SIP/3303-00000008", "1?continue") in new stack
-- Goto (macro-user-callerid,s,18)
-- Executing [[email protected]:18] NoOp("SIP/3303-00000008", "Using CallerID "Admissions 3303" <3303>") in new stack
-- Executing [[email protected]:2] Set("SIP/3303-00000008", "RingGroupMethod=none") in new stack
-- Executing [[email protected]:3] Set("SIP/3303-00000008", "VMBOX=3301") in new stack
-- Executing [[email protected]:4] Set("SIP/3303-00000008", "EXTTOCALL=3301") in new stack
-- Executing [[email protected]:5] Set("SIP/3303-00000008", "CFUEXT=") in new stack
-- Executing [[email protected]:6] Set("SIP/3303-00000008", "CFBEXT=") in new stack
-- Executing [[email protected]:7] Set("SIP/3303-00000008", "RT=15") in new stack
-- Executing [[email protected]:8] Macro("SIP/3303-00000008", "record-enable|3301|IN") in new stack
-- Executing [[email protected]:1] GotoIf("SIP/3303-00000008", "1?check") in new stack

– Goto (macro-record-enable,s,4)
– Executing [[email protected]:4] AGI(“SIP/3303-00000008”, “recordingcheck|20091122-102228|1258903348.8”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20091122-102228|1258903348.8: Inbound recording enabled.
recordingcheck|20091122-102228|1258903348.8: CALLFILENAME=20091122-102228-1258903348.8
– AGI Script recordingcheck completed, returning 0
– Executing [[email protected]:999] MixMonitor(“SIP/3303-00000008”, “20091122-102228-1258903348.8.wav||”) in new stack
– Executing [[email protected]:9] Macro(“SIP/3303-00000008”, “dial|15|trWw|3301”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/3303-00000008”, “1?dial”) in new stack
– Goto (macro-dial,s,3)
– Executing [[email protected]:3] AGI(“SIP/3303-00000008”, “dialparties.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
== Begin MixMonitor Recording SIP/3303-00000008
dialparties.agi: Starting New Dialparties.agi
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager_additional.conf’: Found
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found
== Manager ‘admin’ logged on from 127.0.0.1
dialparties.agi: Caller ID name is ‘Admissions 3303’ number is '3303’
dialparties.agi: USE_CONFIRMATION: 'FALSE’
dialparties.agi: RINGGROUP_INDEX: ''
dialparties.agi: Methodology of ring is ‘none’
– dialparties.agi: Added extension 3301 to extension map
– dialparties.agi: Extension 3301 cf is disabled
– dialparties.agi: Extension 3301 do not disturb is disabled
> dialparties.agi: extnum 3301 has: cw: 1; hascfb: 0 [] hascfu: 0 []
dialparties.agi: ExtensionState: 0
– dialparties.agi: dbset CALLTRACE/3301 to 3303
– dialparties.agi: Filtered ARG3: 3301
– AGI Script dialparties.agi completed, returning 0
– Executing [[email protected]:7] Dial(“SIP/3303-00000008”, “SIP/3301|15|trWw”) in new stack
– Called 3301
== Manager ‘admin’ logged off from 127.0.0.1
– SIP/3301-00000009 is ringing
– SIP/3301-00000009 answered SIP/3303-00000008
– Executing [[email protected]:1] Macro(“SIP/3303-00000008”, “hangupcall”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/3303-00000008”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,4)
– Executing [[email protected]:4] GotoIf(“SIP/3303-00000008”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,7)
– Executing [[email protected]:7] GotoIf(“SIP/3303-00000008”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [[email protected]:9] Hangup(“SIP/3303-00000008”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/3303-00000008’ in macro ‘hangupcall’
== Spawn h extension (macro-dial, h, 1) exited non-zero on ‘SIP/3303-00000008’
== Spawn extension (macro-dial, s, 7) exited non-zero on ‘SIP/3303-00000008’ in macro ‘dial’
== Spawn extension (macro-exten-vm, s, 9) exited non-zero on ‘SIP/3303-00000008’ in macro ‘exten-vm’
== Spawn extension (from-internal, 3301, 1) exited non-zero on ‘SIP/3303-00000008’
== MixMonitor close filestream
== End MixMonitor Recording SIP/3303-00000008

Thanks in advance for your help.

I believe this has already been fixed. Please see this issue:

https://issues.asterisk.org/view.php?id=16314 which is a duplicate of this issue: https://issues.asterisk.org/view.php?id=16268

You can either upgrade to the latest 1.4 branch by checking it out of subversion, or you can wait for the 1.4.28 release (candidates).

Thanks!
Leif Madsen.
Asterisk Bug Marshal.

I will check out a subversion