Speaking Clock Dies (FPBX 2.5.1.1)

General Help
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 [*60@from-internal:1] Answer(“SIP/217-0a0a60a8”, “”) in new stack
– Executing [*60@from-internal:2] Wait(“SIP/217-0a0a60a8”, “1”) in new stack
– Executing [*60@from-internal:3] Set(“SIP/217-0a0a60a8”, “NumLoops=0”) in new stack
– Executing [*60@from-internal:4] Set(“SIP/217-0a0a60a8”, “FutureTime=1232407194”) in new stack
– Executing [*60@from-internal:5] Playback(“SIP/217-0a0a60a8”, “at-tone-time-exactly”) in new stack
– <SIP/217-0a0a60a8> Playing ‘at-tone-time-exactly.ulaw’ (language ‘en’)
– Executing [*60@from-internal:6] GotoIf(“SIP/217-0a0a60a8”, “1?hr24format”) in new stack
– Goto (from-internal,*60,9)
– Executing [*60@from-internal: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 [h@from-internal:1] Macro(“SIP/217-0a0a60a8”, “hangupcall”) in new stack
– Executing [s@macro-hangupcall:1] ResetCDR(“SIP/217-0a0a60a8”, “w”) in new stack
– Executing [s@macro-hangupcall:2] NoCDR(“SIP/217-0a0a60a8”, “”) in new stack
– Executing [s@macro-hangupcall:3] GotoIf(“SIP/217-0a0a60a8”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,6)
– Executing [s@macro-hangupcall:6] GotoIf(“SIP/217-0a0a60a8”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [s@macro-hangupcall:9] GotoIf(“SIP/217-0a0a60a8”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,11)
– Executing [s@macro-hangupcall: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.

2

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).

3

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.

4

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