[Asterisk-Users] Wierd Message

Ricardo Villa ricvil at telesip.net
Tue Aug 5 15:35:25 MST 2003


I have attached the output.  It is just one test call that goes to
voicemail.  You can see the NOTICE message several times.

There is one thing interesting to note.  If I start * from the console
"asterisk -cvvv" on the server I can repreduce it almost always.  But if I
start it from a remote X-Term with the same command I can't seem to
reproduce it.  Could it be an environment variable from the Terminal?  I
looked but nothing seemed obvious.

Regards,
Ricardo Villa


----- Original Message -----
From: "Martin Pycko" <martinp at digium.com>
To: <asterisk-users at lists.digium.com>
Sent: Tuesday, August 05, 2003 4:25 PM
Subject: Re: [Asterisk-Users] Wierd Message


> Can you send a trace from your screen after you turn of the debug in
> /etc/asterisk/logger.conf
>
> console => blabla,debug
>
> regards
> Martin
>
> On Tue, 5 Aug 2003, Ricardo Villa wrote:
>
> > Is it possible to know what application?  The extension I'm daling is
very
> > simple:
> > exten => 1001,1,Dial(SIP/1001,15)
> > exten => 1001,2,Voicemail2(u1001)
> >
> > As soon as the Voicemail picks up the NOTICE line appears multiple times
on
> > the console.
> >
> > Thanks,
> > Ricardo
> >
> > ----- Original Message -----
> > From: "Martin Pycko" <martinp at digium.com>
> > To: <asterisk-users at lists.digium.com>
> > Sent: Tuesday, August 05, 2003 3:57 PM
> > Subject: Re: [Asterisk-Users] Wierd Message
> >
> >
> > > It means that some application scheduled an execution of some routine
in
> > > the past, eg: it will never be executed since it's way in the past ...
> > >
> > > regards
> > > Martin
> > >
> > > On Tue, 5 Aug 2003, Ricardo Villa wrote:
> > >
> > > > Hi,
> > > >
> > > > Whenever someone leaves a Voicemail in our system we get this
message on
> > the
> > > > console:
> > > >
> > > > NOTICE[18447]: File sched.c, Line 209 (sched_settime): Request to
> > schedule
> > > > in the past?!?!
> > > >
> > > > Does anybody know what it means?
> > > >
> > > > Ricardo.
> > > >
> > > > _______________________________________________
> > > > Asterisk-Users mailing list
> > > > Asterisk-Users at lists.digium.com
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > >
> > >
> > > _______________________________________________
> > > Asterisk-Users mailing list
> > > Asterisk-Users at lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
-------------- next part --------------
[root at maui root]# asterisk -cvvv
  == Parsing '/etc/asterisk/asterisk.conf': Found
Asterisk CVS-07/27/03-00:59:20, Copyright (C) 1999-2001 Linux Support Services, Inc.
Written by Mark Spencer <markster at linux-support.net>
=========================================================================
  == Parsing '/etc/asterisk/logger.conf': Found
Asterisk Event Logger Started /var/log/asterisk/event_log
  == Manager registered action Ping
  == Manager registered action Logoff
  == Manager registered action Hangup
  == Manager registered action Status
  == Manager registered action Redirect
  == Manager registered action Originate
  == Manager registered action MailboxStatus
  == Manager registered action Command
  == Manager registered action ExtensionState
  == Manager registered action AbsoluteTimeout
  == Manager registered action MailboxCount
  == Parsing '/etc/asterisk/manager.conf': Found
  == Parsing '/etc/asterisk/rtp.conf': Found
  == RTP Allocating from port range 10000 -> 15000
Asterisk PBX Core Initializing
Registering builtin applications:
 [AbsoluteTimeout]
  == Registered application 'AbsoluteTimeout'
 [Answer]
  == Registered application 'Answer'
 [BackGround]
  == Registered application 'BackGround'
 [Busy]
  == Registered application 'Busy'
 [Congestion]
  == Registered application 'Congestion'
 [DigitTimeout]
  == Registered application 'DigitTimeout'
 [Goto]
  == Registered application 'Goto'
 [GotoIf]
  == Registered application 'GotoIf'
 [GotoIfTime]
  == Registered application 'GotoIfTime'
 [Hangup]
  == Registered application 'Hangup'
 [NoOp]
  == Registered application 'NoOp'
 [Prefix]
  == Registered application 'Prefix'
 [ResponseTimeout]
  == Registered application 'ResponseTimeout'
 [Ringing]
  == Registered application 'Ringing'
 [SayNumber]
  == Registered application 'SayNumber'
 [SayDigits]
  == Registered application 'SayDigits'
 [SetAccount]
  == Registered application 'SetAccount'
 [SetGlobalVar]
  == Registered application 'SetGlobalVar'
 [SetLanguage]
  == Registered application 'SetLanguage'
 [SetVar]
  == Registered application 'SetVar'
 [StripMSD]
  == Registered application 'StripMSD'
 [Suffix]
  == Registered application 'Suffix'
 [Wait]
  == Registered application 'Wait'
Asterisk Dynamic Loader Starting:
  == Parsing '/etc/asterisk/modules.conf': Found
 [chan_modem.so] => (Generic Voice Modem Driver)
  == Parsing '/etc/asterisk/modem.conf': Found
  == Loading modem driver chan_modem_aopen.so => (A/Open (Rockwell Chipset) ITU-2 VoiceModem Driver)
  == Registered channel type 'Modem' (Generic Voice Modem Channel Driver)
 [res_musiconhold.so] => (Music On Hold Resource)
  == Parsing '/etc/asterisk/musiconhold.conf': Found
Warning, flexibel rate not heavily tested!
  == Registered application 'MusicOnHold'
  == Registered application 'WaitMusicOnHold'
  == Registered application 'SetMusicOnHold'
 [res_adsi.so] => (Call Parking Resource)
  == Parsing '/etc/asterisk/adsi.conf': Found
 [res_parking.so] => (Call Parking Resource)
  == Parsing '/etc/asterisk/parking.conf': Found
    -- Registered extension context 'parkedcalls'
    -- Added extension '701' priority 1 to parkedcalls
    -- Added extension '702' priority 1 to parkedcalls
    -- Added extension '703' priority 1 to parkedcalls
    -- Added extension '704' priority 1 to parkedcalls
    -- Added extension '705' priority 1 to parkedcalls
    -- Added extension '706' priority 1 to parkedcalls
    -- Added extension '707' priority 1 to parkedcalls
    -- Added extension '708' priority 1 to parkedcalls
    -- Added extension '709' priority 1 to parkedcalls
    -- Added extension '710' priority 1 to parkedcalls
    -- Added extension '711' priority 1 to parkedcalls
    -- Added extension '712' priority 1 to parkedcalls
    -- Added extension '713' priority 1 to parkedcalls
    -- Added extension '714' priority 1 to parkedcalls
    -- Added extension '715' priority 1 to parkedcalls
    -- Added extension '716' priority 1 to parkedcalls
    -- Added extension '717' priority 1 to parkedcalls
    -- Added extension '718' priority 1 to parkedcalls
    -- Added extension '719' priority 1 to parkedcalls
    -- Added extension '720' priority 1 to parkedcalls
  == Registered application 'ParkedCall'
 [res_crypto.so] => (Cryptographic Digital Signatures)
    -- Loaded PUBLIC key 'iaxtel'
 [res_indications.so] => (Indications Configuration)
  == Parsing '/etc/asterisk/indications.conf': Found
    -- Registered indication country 'us'
    -- Registered indication country 'au'
    -- Registered indication country 'fr'
    -- Registered indication country 'de'
    -- Registered indication country 'nl'
    -- Registered indication country 'uk'
    -- Setting default indication country to 'us'
  == Registered application 'Playtones'
  == Registered application 'StopPlaytones'
 [res_monitor.so] => (Call Monitoring Resource)
  == Registered application 'Monitor'
  == Registered application 'StopMonitor'
  == Registered application 'ChangeMonitor'
  == Manager registered action Monitor
  == Manager registered action StopMonitor
  == Manager registered action ChangeMonitor
 [chan_iax.so] => (Inter Asterisk eXchange)
  == Manager registered action IAXpeers
  == Parsing '/etc/asterisk/iax.conf': Found
  == Registered channel type 'IAX' (Inter Asterisk eXchange Drver)
  == Using TOS bits 16
  == IAX Ready and Listening on 0.0.0.0 port 5036
 [chan_sip.so] => (Session Initiation Protocol (SIP))
  == Parsing '/etc/asterisk/sip.conf': Found
  == SIP Listening on 0.0.0.0:5066
  == Using TOS bits 0
  == Registered channel type 'sip' (Session Initiation Protocol (SIP))
 [chan_oss.so] => (OSS Console Channel Driver)
WARNING[1024]: File chan_oss.c, Line 423 (soundcard_init): Unable to open /dev/dsp: Device or resource busy
  == No sound card detected -- console channel will be unavailable
  == Turn off OSS support by adding 'noload=chan_oss.so' in /etc/asterisk/modules.conf
 [chan_modem_bestdata.so] => (BestData (Conexant V.90 Chipset) VoiceModem Driver)
 [chan_modem_i4l.so] => (ISDN4Linux Emulated Modem Driver)
 [chan_agent.so] => (Agent Proxy Channel)
  == Registered channel type 'Agent' (Call Agent Proxy Channel)
  == Registered application 'AgentLogin'
  == Registered application 'AgentCallbackLogin'
  == Parsing '/etc/asterisk/agents.conf': Found
 [chan_mgcp.so] => (Media Gateway Control Protocol (MGCP))
  == Parsing '/etc/asterisk/mgcp.conf': Found
  == MGCP Listening on 0.0.0.0:2427
  == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP))
 [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2))
  == Manager registered action IAXpeers
  == Parsing '/etc/asterisk/iax.conf': Found
WARNING[1024]: File chan_iax2.c, Line 5061 (set_config): Ignoring port for now
  == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2))
  == Using TOS bits 16
  == IAX Ready and Listening on 0.0.0.0 port 4569
 [chan_local.so] => (Local Proxy Channel)
  == Registered channel type 'Local' (Local Proxy Channel Driver)
 [chan_phone.so] => (Linux Telephony API Support)
  == Parsing '/etc/asterisk/phone.conf': Found
  == Registered channel type 'Phone' (Standard Linux Telephony API Driver)
 [pbx_config.so] => (Text Extension Configuration)
  == Parsing '/etc/asterisk/extensions.conf': Found
    -- Registered extension context 'default'
    -- Added extension '9990' priority 1 to default
    -- Added extension '9990' priority 2 to default
    -- Added extension '9990' priority 3 to default
    -- Added extension '9990' priority 4 to default
    -- Added extension '9990' priority 5 to default
    -- Added extension '9990' priority 6 to default
    -- Added extension '_1010' priority 1 to default
    -- Added extension '_XXXX' priority 1 to default
    -- Added extension '_XXXX' priority 2 to default
    -- Added extension '_XXXX' priority 3 to default
    -- Added extension '_01XXXXXXXXXX' priority 1 to default
    -- Registered extension context 'MainMenu'
    -- Added extension 's' priority 1 to MainMenu
    -- Added extension 's' priority 2 to MainMenu
    -- Added extension 's' priority 3 to MainMenu
    -- Added extension 's' priority 4 to MainMenu
    -- Added extension 's' priority 5 to MainMenu
    -- Added extension 's' priority 6 to MainMenu
    -- Added extension 's' priority 7 to MainMenu
    -- Added extension 's' priority 8 to MainMenu
    -- Added extension '1' priority 1 to MainMenu
    -- Added extension '2' priority 1 to MainMenu
    -- Added extension '3' priority 1 to MainMenu
    -- Registered extension context 'sip'
    -- Added extension '1001' priority 1 to sip
    -- Added extension '1001' priority 2 to sip
    -- Added extension '1002' priority 1 to sip
    -- Added extension '1002' priority 2 to sip
    -- Added extension '1002' priority 3 to sip
    -- Added extension '1010' priority 1 to sip
    -- Added extension '1010' priority 2 to sip
    -- Added extension '9998' priority 1 to sip
    -- Added extension '9999' priority 1 to sip
    -- Added extension '_XXXX' priority 1 to sip
    -- Added extension '_01XXXXXXXXXX' priority 1 to sip
 [pbx_wilcalu.so] => (Wil Cal U (Auto Dialer))
Entered Wil-Calu fd=16
 [pbx_spool.so] => (Outgoing Spool Support)
/var/spool/asterisk/outgoing
 [skipping pbx_gtkconsole.so]
 [app_dial.so] => (Dialing Application)
  == Registered application 'Dial'
 [app_playback.so] => (Trivial Playback Application)
  == Registered application 'Playback'
 [app_voicemail.so] => (Comedian Mail (Voicemail System))
  == Registered application 'VoiceMail'
  == Registered application 'VoiceMailMain'
 [app_directory.so] => (Extension Directory)
  == Registered application 'Directory'
 [skipping app_intercom.so]
 [app_mp3.so] => (Silly MP3 Application)
  == Registered application 'MP3Player'
 [app_system.so] => (Generic System() application)
  == Registered application 'System'
 [app_echo.so] => (Simple Echo Application)
  == Registered application 'Echo'
 [app_record.so] => (Trivial Record Application)
  == Registered application 'Record'
 [app_image.so] => (Image Transmission Application)
  == Registered application 'SendImage'
 [app_url.so] => (Send URL Applications)
  == Registered application 'SendURL'
 [app_disa.so] => (DISA (Direct Inward System Access) Application)
  == Registered application 'DISA'
 [app_agi.so] => (Asterisk Gateway Interface (AGI))
  == Registered application 'EAGI'
  == Registered application 'AGI'
 [app_qcall.so] => (Call from Queue)
 [app_adsiprog.so] => (Asterisk ADSI Programming Application)
  == Registered application 'ADSIProg'
 [app_getcpeid.so] => (Get ADSI CPE ID)
  == Registered application 'GetCPEID'
 [app_milliwatt.so] => (Digital Milliwatt (mu-law) Test Application)
  == Registered application 'Milliwatt'
 [app_zapateller.so] => (Block Telemarketers with Special Information Tone)
  == Registered application 'Zapateller'
 [app_datetime.so] => (Date and Time)
  == Registered application 'DateTime'
 [app_setcallerid.so] => (Set CallerID Application)
  == Registered application 'SetCallerID'
 [app_festival.so] => (Simple Festival Interface)
  == Registered application 'Festival'
 [app_queue.so] => (True Call Queueing)
  == Registered application 'Queue'
  == Manager registered action Queues
  == Manager registered action QueueStatus
  == Registered application 'AddQueueMember'
  == Registered application 'RemoveQueueMember'
  == Parsing '/etc/asterisk/queues.conf': Found
 [app_senddtmf.so] => (Send DTMF digits Application)
  == Registered application 'SendDTMF'
 [app_parkandannounce.so] => (Call Parking and Announce Application)
  == Registered application 'ParkAndAnnounce'
 [app_striplsd.so] => (Strip trailing digits)
  == Registered application 'StripLSD'
 [app_setcidname.so] => (Set CallerID Name)
  == Registered application 'SetCIDName'
 [app_lookupcidname.so] => (Look up CallerID Name from local database)
  == Registered application 'LookupCIDName'
 [app_substring.so] => (Save substring digits in a given variable)
  == Registered application 'SubString'
 [app_macro.so] => (Extension Macros)
  == Registered application 'Macro'
 [app_authenticate.so] => (Authentication Application)
  == Registered application 'Authenticate'
 [app_softhangup.so] => (Hangs up the requested channel)
  == Registered application 'SoftHangup'
 [app_lookupblacklist.so] => (Look up Caller*ID name/number from blacklist database)
  == Registered application 'LookupBlacklist'
 [app_waitforring.so] => (Waits until first ring after time)
  == Registered application 'WaitForRing'
 [app_privacy.so] => (Require phone number to be entered, if no CallerID sent)
  == Registered application 'PrivacyManager'
 [app_db.so] => (Database access functions for Asterisk extension logic)
  == Registered application 'DBget'
  == Registered application 'DBput'
  == Registered application 'DBdel'
  == Registered application 'DBdeltree'
 [app_chanisavail.so] => (Check if channel is available)
  == Registered application 'ChanIsAvail'
 [app_enumlookup.so] => (ENUM Lookup)
  == Registered application 'EnumLookup'
 [app_voicemail2.so] => (Comedian Mail (Voicemail System))
  == Parsing '/etc/asterisk/voicemail.conf': Found
  == Registered application 'VoiceMail2'
  == Registered application 'VoiceMailMain2'
 [app_transfer.so] => (Transfer)
  == Registered application 'Transfer'
 [codec_ilbc.so] => (iLBC/PCM16 (signed linear) Codec Translator)
  == Registered translator 'ilbctolin' from format 10 to 6, cost 12
  == Registered translator 'lintoilbc' from format 6 to 10, cost 66
 [codec_gsm.so] => (GSM/PCM16 (signed linear) Codec Translator)
  == Registered translator 'gsmtolin' from format 1 to 6, cost 7
  == Registered translator 'lintogsm' from format 6 to 1, cost 10
 [codec_mp3_d.so] => (MP3/PCM16 (signed linear) Translator (Decoder only))
  == Registered translator 'mp3tolin' from format 4 to 6, cost 39
 [codec_lpc10.so] => (LPC10 2.4kbps (signed linear) Voice Coder)
  == Registered translator 'lpc10tolin' from format 7 to 6, cost 10
  == Registered translator 'lintolpc10' from format 6 to 7, cost 13
 [codec_adpcm.so] => (Adaptive Differential PCM Coder/Decoder)
  == Registered translator 'adpcmtolin' from format 5 to 6, cost 1
  == Registered translator 'lintoadpcm' from format 6 to 5, cost 1
 [codec_ulaw.so] => (Mu-law Coder/Decoder)
  == Registered translator 'ulawtolin' from format 2 to 6, cost 1
  == Registered translator 'lintoulaw' from format 6 to 2, cost 1
 [codec_alaw.so] => (A-law Coder/Decoder)
  == Registered translator 'alawtolin' from format 3 to 6, cost 1
  == Registered translator 'lintoalaw' from format 6 to 3, cost 1
 [codec_a_mu.so] => (A-law and Mulaw direct Coder/Decoder)
  == Registered translator 'alawtoulaw' from format 3 to 2, cost 1
  == Registered translator 'ulawtoalaw' from format 2 to 3, cost 1
 [format_gsm.so] => (Raw GSM data)
  == Registered file format gsm, extension(s) gsm
 [format_wav.so] => (Microsoft WAV format (8000hz Signed Linear))
  == Registered file format wav, extension(s) wav
 [format_mp3.so] => (MPEG-1,2 Layer 3 File Format Support)
  == Registered file format mp3, extension(s) mp3|mpeg3
 [format_wav_gsm.so] => (Microsoft WAV format (Proprietary GSM))
  == Registered file format wav49, extension(s) WAV
 [format_vox.so] => (Dialogic VOX (ADPCM) File Format)
  == Registered file format vox, extension(s) vox
 [format_pcm.so] => (Raw uLaw 8khz Audio support (PCM))
  == Registered file format pcm, extension(s) pcm|ulaw|ul|mu
 [format_g729.so] => (Raw G729 data)
  == Registered file format g729, extension(s) g729
 [format_pcm_alaw.so] => (Raw aLaw 8khz PCM Audio support)
  == Registered file format alaw, extension(s) alaw|al
 [format_h263.so] => (Raw h263 data)
  == Registered file format h263, extension(s) h263
 [format_jpeg.so] => (JPEG (Joint Picture Experts Group) Image Format)
  == Registered format 'jpg' (JPEG (Joint Picture Experts Group))
 [cdr_csv.so] => (Comma Separated Values CDR Backend)
 [codec_g729b.so] => (Annex B (floating point) G.729/PCM16 Codec Translator)
  == Detected 2 licensed G.729 transcoders
WARNING[1024]: File translate.c, Line 218 (calc_cost): Translator 'g729tolinb' does not produce sample frames.
  == Registered translator 'g729tolinb' from format 8 to 6, cost 99999
  == Registered translator 'lintog729b' from format 6 to 8, cost 75
 [cdr_mysql.so] => (MySQL CDR Backend)
  == Parsing '/etc/asterisk/cdr_mysql.conf': Found
  == Parsing '/etc/asterisk/enum.conf': Found
Asterisk Ready.
*CLI> DEBUG[6151]: File chan_sip.c, Line 3611 (check_user): Setting NAT on RTP to 0
DEBUG[6151]: File chan_sip.c, Line 923 (find_user): Call from user '1002' is 1 out of 0
DEBUG[6151]: File chan_sip.c, Line 3118 (build_route): build_route: Contact hop: <sip:1002 at 192.168.0.15:5060;user=phone;transport=udp>
    -- Executing Dial("SIP/1002-9e8f", "SIP/1001|15") in new stack
DEBUG[13326]: File chan_sip.c, Line 618 (create_addr): Setting NAT on RTP to 0
NOTICE[13326]: File app_dial.c, Line 495 (dial_exec): Unable to create channel of type 'SIP'
  == Everyone is busy at this time
    -- Executing VoiceMail2("SIP/1002-9e8f", "u1001") in new stack
DEBUG[13326]: File app_voicemail2.c, Line 995 (leave_voicemail): voicemail/default/1001/unavail doesn't exist, doing what we can
DEBUG[13326]: File rtp.c, Line 982 (ast_rtp_write): Ooh, format changed from 0 to 256
    -- Playing 'vm-theperson'
DEBUG[6151]: File chan_sip.c, Line 529 (__sip_ack): Stopping retransmission on '3645781250 at 192.168.0.15' of Response 1: Found
    -- Playing 'digits/1'
    -- Playing 'digits/0'
    -- Playing 'digits/0'
NOTICE[13326]: File sched.c, Line 209 (sched_settime): Request to schedule in the past?!?!
    -- Playing 'digits/1'
NOTICE[13326]: File sched.c, Line 209 (sched_settime): Request to schedule in the past?!?!
    -- Playing 'vm-isunavail'
    -- Playing 'vm-intro'
NOTICE[13326]: File sched.c, Line 209 (sched_settime): Request to schedule in the past?!?!
    -- Playing 'beep'
DEBUG[13326]: File app_voicemail2.c, Line 766 (play_and_record): play_and_record: <None>, /var/spool/asterisk/voicemail/default/1001/INBOX/msg0010, 'gsm'
DEBUG[13326]: File app_voicemail2.c, Line 783 (play_and_record): Recording Formats: sfmts=gsm
    -- x=0, open writing:  /var/spool/asterisk/voicemail/default/1001/INBOX/msg0010 format: gsm, 0x80cb578
    -- User hung up
DEBUG[13326]: File app_voicemail2.c, Line 579 (sendmail): Attaching file '/var/spool/asterisk/voicemail/default/1001/INBOX/msg0010', format 'gsm', uservm is '-1', global is -1
  == Spawn extension (sip, 1001, 2) exited non-zero on 'SIP/1002-9e8f'
DEBUG[13326]: File chan_sip.c, Line 951 (sip_hangup): find_user(1002)



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