Speaking Clock Dies (FPBX 2.5.1.1)

I use *60 to test phones, and after building my new system, I am having a problem right off the bat.

Dialing *60 I get “at the sound of the tone the time will be exactly 16:19 --” hangup

Here is a log
Connected to Asterisk 1.6.0.3 currently running on asterisk (pid = 6505)
Verbosity is at least 3
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [*[email protected]:1] Answer(“SIP/217-0a0a60a8”, “”) in new stack
– Executing [*[email protected]:2] Wait(“SIP/217-0a0a60a8”, “1”) in new stack
– Executing [*[email protected]:3] Set(“SIP/217-0a0a60a8”, “NumLoops=0”) in new stack
– Executing [*[email protected]:4] Set(“SIP/217-0a0a60a8”, “FutureTime=1232407194”) in new stack
– Executing [*[email protected]:5] Playback(“SIP/217-0a0a60a8”, “at-tone-time-exactly”) in new stack
– <SIP/217-0a0a60a8> Playing ‘at-tone-time-exactly.ulaw’ (language ‘en’)
– Executing [*[email protected]:6] GotoIf(“SIP/217-0a0a60a8”, “1?hr24format”) in new stack
– Goto (from-internal,*60,9)
– Executing [*[email protected]:9] SayUnixTime(“SIP/217-0a0a60a8”, “1232407194,kM ‘and’ S ‘seconds’”) in new stack
– <SIP/217-0a0a60a8> Playing ‘digits/16.ulaw’ (language ‘en’)
– <SIP/217-0a0a60a8> Playing ‘digits/19.ulaw’ (language ‘en’)
== Spawn extension (from-internal, *60, 9) exited non-zero on ‘SIP/217-0a0a60a8’
– Executing [[email protected]:1] Macro(“SIP/217-0a0a60a8”, “hangupcall”) in new stack
– Executing [[email protected]:1] ResetCDR(“SIP/217-0a0a60a8”, “w”) in new stack
– Executing [[email protected]:2] NoCDR(“SIP/217-0a0a60a8”, “”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/217-0a0a60a8”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,6)
– Executing [[email protected]:6] GotoIf(“SIP/217-0a0a60a8”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [[email protected]:9] GotoIf(“SIP/217-0a0a60a8”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,11)
– Executing [[email protected]:11] Hangup(“SIP/217-0a0a60a8”, “e[0;37;40 m”) in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on ‘SIP/217-0a0a60a8’ in macro ‘hangupcall’
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on ‘SIP/217-0a0a60a8’

The speaking clock is not an necessity for me, but it does indicate a problem that I might have with other features.

The new system was built following the guide here fairly closely.
http://www.powerpbx.org/content/centos-asterisk-freepbx-install-guide-centos-v5x-asterisk-v16x-freepbx-v24x

I can use voicemail no problem but haven’t tried anything else yet.
My Wireshark shows the server issuing a SIP BYE to the phone.

Could be something as simple as a missing sound file for the beep, or perhaps the wrong permissions or ownership on that particular file (or the directory in which it resides).

I’ve checked permissions, they are properly set. I had my soundfiles in the wrong directory on a 1.4 install once, and although it didn’t play the sounds (obviously) it didn’t hangup the call either.

I found in my trixbox that SayUnixTime has some hardcoded soundfiles names/path in function ast_say_date_with_format_* in the file say.c
In my case i experienced the same hangup if one of those was missing, maybe it is worth checking it (even if you use a different version).

see: http://trixbox.org/forums/trixbox-forums/help/speaking-clock-german-hangup-extension-exited-non-zero