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trixbox speaking clock now has no seconds or beep

johnnysme

trixbox speaking clock now has no seconds or beep
« on: May 07, 2007, 03:04:59 PM »
i have smeserver-trixbox-fws-beta2 installed & upgraded as per
http://forums.contribs.org/index.php?topic=34264.0
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all has been well but today my speaking clock no longer reports the seconds or the beep it just goes to the hang up bit after the minutes have been spoken here is a log of the output

 asterisk -rvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv
vvvvvvv
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.2.13 svn rev 47264, Copyright (C) 1999 - 2006 Digium, Inc. and others
.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'show license' for details.
=========================================================================
Connected to Asterisk 1.2.13 svn rev 47264 currently running on asterisk1 (pid =
 4634)
Verbosity was 1 and is now 56
    -- Executing Answer("SIP/201-094105e8", "") in new stack
    -- Executing Wait("SIP/201-094105e8", "1") in new stack
    -- Executing Set("SIP/201-094105e8", "NumLoops=0") in new stack
    -- Executing Set("SIP/201-094105e8", "FutureTime=1178539855") in new stack
    -- Executing Playback("SIP/201-094105e8", "at-tone-time-exactly") in new sta
ck
    -- Playing 'at-tone-time-exactly' (language 'en')
    -- Executing GotoIf("SIP/201-094105e8", "1?hr24format") in new stack
    -- Goto (from-internal,*60,9)
    -- Executing SayUnixTime("SIP/201-094105e8", "1178539855||kM \'and\' S \'sec
onds\'") in new stack
    -- Playing 'digits/13' (language 'en')
    -- Playing 'digits/10' (language 'en')
  == Spawn extension (from-internal, *60, 9) exited non-zero on 'SIP/201-094105e
8'
    -- Executing Macro("SIP/201-094105e8", "hangupcall") in new stack
    -- Executing ResetCDR("SIP/201-094105e8", "w") in new stack
    -- Executing NoCDR("SIP/201-094105e8", "") in new stack
    -- Executing GotoIf("SIP/201-094105e8", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,6)
    -- Executing GotoIf("SIP/201-094105e8", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing Wait("SIP/201-094105e8", "5") in new stack
    -- Executing Hangup("SIP/201-094105e8", "") in new stack
  == Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'SIP/201-09410
5e8' in macro 'hangupcall'
  == Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'SIP/201-09410
5e8'
asterisk1*CLI> exit
Executing last minute cleanups


here is my the excert from extensions_additional.conf


[app-speakingclock]
include => app-speakingclock-custom
exten => *60,1,Answer
exten => *60,n,Wait(1)
exten => *60,n,Set(NumLoops=0)
exten => *60,n(start),Set(FutureTime=$[${EPOCH} + 11])
exten => *60,n,Playback(at-tone-time-exactly)
exten => *60,n,GotoIf($[\"${TIMEFORMAT}\" = \"kM\"]?hr24format)
exten => *60,n,SayUnixTime(${FutureTime},,IM \\\'and\\\' S \\\'seconds\\\' p)
exten => *60,n,Goto(waitloop)
exten => *60,n(hr24format),SayUnixTime(${FutureTime},,kM \\\'and\\\' S \\\'seconds\\\')
exten => *60,n(waitloop),Set(TimeLeft=$[${FutureTime} - ${EPOCH}])
exten => *60,n,GotoIf($[${TimeLeft} < 1]?playbeep)
exten => *60,n,Wait(1)
exten => *60,n,Goto(waitloop)
exten => *60,n(playbeep),Playback(beep)
exten => *60,n,Wait(5)
exten => *60,n,Set(NumLoops=$[${NumLoops} + 1])
exten => *60,n,GotoIf($[${NumLoops} < 5]?start)
exten => *60,n,Playback(goodbye)
exten => *60,n,Hangup
; end of [app-speakingclock]


i have another sme 7.1.3 with the same smeserver-trixbox-fws-beta2 installed & upgraded it plays the speaking clock correctly, debug output below

[root@tb1 ~]# asterisk -rvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.2.13 svn rev 47264, Copyright (C) 1999 - 2006 Digium, Inc. and others
.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'show license' for details.
=========================================================================
Connected to Asterisk 1.2.13 svn rev 47264 currently running on tb1 (pid = 4053)
Verbosity was 1 and is now 42
    -- Unregistered SIP '203'
    -- Registered SIP '203' at 90.0.0.201 port 5066 expires 900
    -- Saved useragent "Grandstream GXP2000 1.1.1.14" for peer 203
    -- Executing Answer("SIP/203-0912d378", "") in new stack
    -- Executing Wait("SIP/203-0912d378", "1") in new stack
    -- Executing Set("SIP/203-0912d378", "NumLoops=0") in new stack
    -- Executing Set("SIP/203-0912d378", "FutureTime=1178540984") in new stack
    -- Executing Playback("SIP/203-0912d378", "at-tone-time-exactly") in new sta
ck
    -- Playing 'at-tone-time-exactly' (language 'en')
    -- Executing GotoIf("SIP/203-0912d378", "1?hr24format") in new stack
    -- Goto (from-internal,*60,9)
    -- Executing SayUnixTime("SIP/203-0912d378", "1178540984||kM 'and' S 'second
s'") in new stack
    -- Playing 'digits/13' (language 'en')
    -- Playing 'digits/20' (language 'en')
    -- Playing 'digits/9' (language 'en')
    -- Playing 'and' (language 'en')
    -- Playing 'digits/40' (language 'en')
    -- Playing 'digits/4' (language 'en')
    -- Playing 'seconds' (language 'en')
    -- Executing Set("SIP/203-0912d378", "TimeLeft=3") in new stack
    -- Executing GotoIf("SIP/203-0912d378", "0?playbeep") in new stack
    -- Executing Wait("SIP/203-0912d378", "1") in new stack
    -- Executing Goto("SIP/203-0912d378", "waitloop") in new stack
    -- Goto (from-internal,*60,10)
    -- Executing Set("SIP/203-0912d378", "TimeLeft=2") in new stack
    -- Executing GotoIf("SIP/203-0912d378", "0?playbeep") in new stack
    -- Executing Wait("SIP/203-0912d378", "1") in new stack
    -- Executing Goto("SIP/203-0912d378", "waitloop") in new stack
    -- Goto (from-internal,*60,10)
    -- Executing Set("SIP/203-0912d378", "TimeLeft=1") in new stack
    -- Executing GotoIf("SIP/203-0912d378", "0?playbeep") in new stack
    -- Executing Wait("SIP/203-0912d378", "1") in new stack
    -- Executing Goto("SIP/203-0912d378", "waitloop") in new stack
    -- Goto (from-internal,*60,10)
    -- Executing Set("SIP/203-0912d378", "TimeLeft=0") in new stack
    -- Executing GotoIf("SIP/203-0912d378", "1?playbeep") in new stack
    -- Goto (from-internal,*60,14)
    -- Executing Playback("SIP/203-0912d378", "beep") in new stack
    -- Playing 'beep' (language 'en')
    -- Executing Wait("SIP/203-0912d378", "5") in new stack
  == Spawn extension (from-internal, *60, 15) exited non-zero on 'SIP/203-0912d3
78'
    -- Executing Macro("SIP/203-0912d378", "hangupcall") in new stack
    -- Executing ResetCDR("SIP/203-0912d378", "w") in new stack
    -- Executing NoCDR("SIP/203-0912d378", "") in new stack
    -- Executing GotoIf("SIP/203-0912d378", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,6)
    -- Executing GotoIf("SIP/203-0912d378", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing Wait("SIP/203-0912d378", "5") in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/203-0912d3
78' in macro 'hangupcall'
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/203-0912d3
78'
tb1*CLI> exit
Executing last minute cleanups
[root@tb1 ~]#

its extensions_additional.conf looks like this
[app-speakingclock]
include => app-speakingclock-custom
exten => *60,1,Answer
exten => *60,n,Wait(1)
exten => *60,n,Set(NumLoops=0)
exten => *60,n(start),Set(FutureTime=$[${EPOCH} + 11])
exten => *60,n,Playback(at-tone-time-exactly)
exten => *60,n,GotoIf($["${TIMEFORMAT}" = "kM"]?hr24format)
exten => *60,n,SayUnixTime(${FutureTime},,IM \'and\' S \'seconds\' p)
exten => *60,n,Goto(waitloop)
exten => *60,n(hr24format),SayUnixTime(${FutureTime},,kM \'and\' S \'seconds\')
exten => *60,n(waitloop),Set(TimeLeft=$[${FutureTime} - ${EPOCH}])
exten => *60,n,GotoIf($[${TimeLeft} < 1]?playbeep)
exten => *60,n,Wait(1)
exten => *60,n,Goto(waitloop)
exten => *60,n(playbeep),Playback(beep)
exten => *60,n,Wait(5)
exten => *60,n,Set(NumLoops=$[${NumLoops} + 1])
exten => *60,n,GotoIf($[${NumLoops} < 5]?start)
exten => *60,n,Playback(goodbye)
exten => *60,n,Hangup

; end of [app-speakingclock]

which is not quite the same as the broken one,
the broken one now has the speaking lines
exten => *60,n,SayUnixTime(${FutureTime},,IM \\\'and\\\' S \\\'seconds\\\' p)
and
exten => *60,n(hr24format),SayUnixTime(${FutureTime},,kM \\\'and\\\' S \\\'seconds\\\')
which have three \\\'s in them

my question is how did these get here, is there a bug in freepbx

when i manually edit these out all is well again but i'm just wondering what put them there in the first place.... i know it wasn't me!!!

ta