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Summary:ASTERISK-01770: app_voicemail.c error as of today's cvs update
Reporter:k0fcc (k0fcc)Labels:
Date Opened:2004-06-06 13:01:12Date Closed:2011-06-07 14:05:08
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) sipdebug.txt
Description:With sip.conf containing

disallow=all
allow=ulaw
allow=alaw
allow=gsm

and the recent cvs update (today), we are now getting an error that wasn't there this morning.

Destroying call '59437aa03f7cc0fa76c962861744f533@172.16.2.150'
  -- Playing 'beep' (language 'en')
  -- Recording the message
  -- x=0, open writing:  /var/spool/asterisk/voicemail/default/100/INBOX/msg0003 format: wav49, 0x81c70c8
  -- x=1, open writing:  /var/spool/asterisk/voicemail/default/100/INBOX/msg0003 format: gsm, 0x81c71e8
  -- x=2, open writing:  /var/spool/asterisk/voicemail/default/100/INBOX/msg0003 format: wav, 0x818c5a0
Jun  6 13:23:42 WARNING[245776]: app_voicemail.c:1377 play_and_record: No audio available on SIP/101-db87??
  -- User hung up
  -- Executing Playback("SIP/101-db87", "something-terribly-wrong") in new stack
  -- Playing 'something-terribly-wrong' (language 'en')
Jun  6 13:23:42 NOTICE[245776]: sched.c:218 sched_settime: Request to schedule in the past?!?!
  -- Executing Playback("SIP/101-db87", "please-contact-tech-supt") in new stack
  -- Playing 'please-contact-tech-supt' (language 'en')
  -- Executing Hangup("SIP/101-db87", "") in new stack
== Spawn extension (macro-stdexten, s, 105) exited non-zero on 'SIP/101-db87' in macro 'stdexten'
== Spawn extension (sip, 100, 1) exited non-zero on 'SIP/101-db87'
  -- Executing Hangup("SIP/101-db87", "") in new stack
== Spawn extension (sip, h, 1) exited non-zero on 'SIP/101-db87'
11 headers, 2 lines


****** STEPS TO REPRODUCE ******

Any attempt to leave voicemail on any phone will produce the message.
Comments:By: Brian West (bkw918) 2004-06-06 13:42:51

This isn't a bug with app_voicemail.. its a config problem on your setup.
By: k0fcc (k0fcc) 2004-06-06 14:00:40

Can you be more specific? It only started  happening after today's update to cvshead.  We've checked all the forums and static information we could find with no success.  Voicemail is actually left but we still encounter this error.  This is a production system.

edited on: 06-06-04 13:45
By: Mark Spencer (markster) 2004-06-06 15:51:48

This sounds like a BYE not making its way and voicemail finally hanging up because no RTP traffic is being received.  Perhaps you can supply a SIP DEBUG.
By: k0fcc (k0fcc) 2004-06-06 16:38:59

The text I pasted in was from a "sip debug peer 101" but I just uploaded the file with several thousand lines removed before and during the conversation. The call is from ext 101 to ext 100 (which then dials 100 & 101). It doesn't matter what extension and who is calling who.

I can also paste in some sanitized conf files if you think it would be relavent.

It would be helpful to know what exactly the no audio msg means.  We get audio on the calls.  

We're using this stdexten modification:

[macro-stdexten];
;
; Standard extension macro:
;   ${ARG1} - Extension  (we could have used ${MACRO_EXTEN} here as well
;   ${ARG2} - Device(s) to ring
;
;exten => s,1,Playback(pls-wait-connect-call)
exten => s,1,Dial(${ARG2},20,TtrH)   ; Ring the interface, 20 seconds maximum
exten => s,2,Voicemail(u${ARG1}) ; If unavailable, send to voicemail w/ unavail announce
exten => s,3,Hangup
;priority of dial above + 101 = start of busy e.g. s,2,Dial + 101 = s,103,Voicemail
exten => s,102,Voicemail(b${ARG1}) ; If busy, send to voicemail w/ busy announce
exten => s,103,Playback(something-terribly-wrong)
exten => s,104,Playback(please-contact-tech-supt)
exten => s,105,Hangup


I added 103 & 104 to catch errors early in the development of this box.


For the particular debug file, these are the two extensions:

;100 = Joey's Sipura
exten => 100,1,Macro(stdexten,100,SIP/100&SIP/101)
;101 = Joey's WISIP
exten => 101,1,Macro(stdexten,101,SIP/101)

and this is the relevant sip.conf

;
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = 172.16.2.150 ; Address to bind to
context = sip ; Default for incoming calls
;srvlookup = yes ; Enable SRV lookups on outbound calls
;pedantic = yes ; Enable slow, pedantic checking for Pingtel
;tos=lowdelay
tos=0x17
maxexpirey=3600 ; Max length of incoming registration we allow
defaultexpirey=120 ; Default length of incoming/outoing registration
;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY
;videosupport=yes ; Turn on support for SIP video
disallow=all ; Disallow all codecs
allow=ulaw
allow=alaw ; Allow codecs in order of preference
allow=gsm
qualify=yes
;allow=ilbc

;
; **** Joey's Entries ****

[100]
type=friend
secret=<secret password>
host=dynamic
;dtmfmode=inband                ; Choices are inband, rfc2833, or info
;defaultip=192.168.0.59
mailbox=100          ; Mailbox for message waiting indicator
context=sip
callerid= "Joey Stanford" <17476697121>
nat=yes
canreinvite=no
qualify=yes

[101]
type=friend
secret=<secret password>
host=dynamic
;dtmfmode=inband                ; Choices are inband, rfc2833, or info
;defaultip=192.168.0.59
mailbox=100          ; Mailbox for message waiting indicator
context=sip
callerid= "Joey Stanford (WIFI)" <17476697121>
nat=yes
canreinvite=no
qualify=yes



and voicemail.conf is straight forward:

100 => <secret Password>,Joey Stanford,joey@stan4d.net,,tz=mountain


Using asterisk -vvvvvvvvvvvvc shows no startup errors.
By: Mark Spencer (markster) 2004-06-06 17:15:18

Does this message show itself *every* time that extension leaves voicemail?

Something is seriously wrong with your SIP device or how it's seeing/not seeing Asterisk.  You notice in the attached debug it's not responding to our 200 OK to its INVITE.
By: k0fcc (k0fcc) 2004-06-06 17:21:30

Mark,

It shows that message for all of our sipphones.  Sipuras, Budgetones, and WISIPs.

It doesn't matter who calls who. It's a pervasive issue. It started this morning with CVSHEAD.
By: Mark Spencer (markster) 2004-06-06 21:08:56

Okay lets get a SIP debug from a phone that *is* responding to the 200 OK and work from there.
By: Mark Spencer (markster) 2004-06-06 22:56:32

You probably just need to get on IRC (irc.freenode.net, #asterisk-bugs) and see if you can find someone to help you go through your config and see what's going on.
By: Mark Spencer (markster) 2004-06-08 16:57:37

What's the story here, did you find anyone on IRC that was helpful?
By: k0fcc (k0fcc) 2004-06-08 18:18:56

We resolved this today.  It was a problem with MEPIS linux apparently.  We did an apt-get upgrade today and this problem went away.  Thanks for the help!