[RESOLVED] voicemail problems with old messages

Hi, I’ve configured voicemail. I can post a message, then listen to it, but if I leave old message and try to login to voicemailmain i have smth like this:
In phone:
You’ve 2 new and …here the connection is lost

in console:
– Executing [999@mojsip:1] VoiceMailMain(“SIP/101-081c3510”, “9100@poczta|s”) in new stack
– Playing ‘vm-youhave’ (language ‘de’)
– Playing ‘digits/2’ (language ‘de’)
– Playing ‘vm-INBOX’ (language ‘de’)
– Playing ‘vm-and’ (language ‘de’)
Aug 14 14:37:44 WARNING[6631]: file.c:557 ast_openstream_full: File digits/1F does not exist in any format
Aug 14 14:37:44 WARNING[6631]: file.c:810 ast_streamfile: Unable to open digits/1F (format 0x4 (ulaw)): No such file or directory
== Spawn extension (mojsip, 999, 1) exited non-zero on ‘SIP/101-081c3510’

conf files :

;The intro can be customized on a per-context basis
1234 => 5678,Company2 User,root@localhost
9100 => 997,Imie,imie@a.aa
101 => 101, Imie2, imie2@a.aa

exten => 100,1,dial(sip/xlite1|20)
exten => 100,2,voicemail(9100@poczta)
;exten => 100,3,voicemail(b100)
exten => 100,3,HangUp

exten => 101,1,dial(sip/101|20)
exten => 101,2,voicemail(101@poczta)
;exten => 101,3,voicemail(b100)
exten => 101,3,HangUp

;exten => 999,1,VoiceMailMain(${CALLERID(num)}@poczta|s)
exten => 999,1,VoiceMailMain(9100@poczta|s)
exten => 999,2,HangUp

; Turn off silence suppression in X-Lite (“Transmit Silence”=YES)!
; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
secret=password ; haslo userka
;regexten=1234 ; When they register, create extension 1234
callerid=“SoftPhone” <100>
host=dynamic ; This device needs to register
;defaultip= ; hmmm moze zadziala
;nat=yes ; X-Lite is behind a NAT router
;canreinvite=no ; Typically set to NO if behind NAT
allow=gsm ; GSM consumes far less bandwidth than ulaw
;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
context=mojsip ; moze bedzie dobry

type=friend ; Friends place calls and receive calls
context=mojsip ;zmienilem z from-sip Context for incoming calls from this user
callerid=“HardPhone” <101>
subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
language=de ; Use German prompts for this user
host=dynamic ; This peer register with us
dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
;defaultip= ; IP used until peer registers
mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
subscribemwi=yes ; Only send notifications if this phone
; subscribes for mailbox notification
vmexten=voicemail ; dialplan extension to reach mailbox
; sets the Message-Account in the MWI notify message
; defaults to global vmexten which defaults to "asterisk"
allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!

Thx for help :smile:

Asterisk is trying to play the sound file for digit 1 and it cant find the file. Something must of happend to it. Try to reinstall asterisk-sounds and see if it works.

i read it that there are 1f messages waiting (31), and Asterisk isn’t converting this to decimal before trying to play the file.

now to go off and record more than 10 messages to see what happens !

ok, scratch that ! i freaked my ATA out by adding 31 VMs and Asterisk read out the number correctly.

if it’s a typo then Dovids answer would be the best starting place.

baconbuttie could explain a bit clearer what do you mean? i don’t understand you :frowning:

me too !

i’ve just re-read the OP and have realised it’s the saved/old (?) messages that might be causing the issue.

i can see that Asterisk is trying to play file ‘1F’ … and why would it be trying to do that ? is it because there are 31 old/saved messages in the mailbox ? perhaps you could have a look in the appropriate vm mailbox and let us know what’s there ?

well i recorded 3 messages, listened to one …so now there are 2 new and 1 old…(then disconnected without deleting the old one). when i try to get into voicemail i have this kind of situation

I’ve reinstalled asterisk-sounds but no effects :frowning:

any help? :frowning:

access VM again now, and post the log fragment. i’m intrigued to see if it’s still trying to get digits/1F.

what language do you have Asterisk running in ?

access vm in what way? i’ve already pasted the log from asterisk console

asterisk is running in mandriva 2k6 polish language

[quote=“rymoholiko”]Aug 14 14:37:44 WARNING[6631]: file.c:557 ast_openstream_full: File digits/1F does not exist in any format
Aug 14 14:37:44 WARNING[6631]: file.c:810 ast_streamfile: Unable to open digits/1F (format 0x4 (ulaw)): No such file or directory

i’m merely trying to establish why Asterisk wants to play the sound file ‘digits/1F’

if it’s not a typo, then you need that file to be present with the correct permissions. what it’s supposed to contain is up to you, as i have no idea how many old messages you have.

maybe it’s a bug with the polish language options. maybe you should go report it as a bug on bugs.digium.com ? maybe a google search would help ?

edit : another quick google, and ip-phone-forum.de/showthread.php?t=67680 would suggest you just need to copy 1.gsm to 1F.gsm in your digits directory.

Thank you very much. Making a copy of 1.gsm with name 1F.gsm fixed the problem. :smile: