Problem with voicemail

Asterisk Asterisk General
1

hello forum, I have a problem with the voicemail, when I want to listen to the messages, says to me that I have new messages, but one cuts the calls, and I review log of asterisk and it shows me this error

– Remote UNIX connection
– Executing [8000@hotel:1] VoiceMailMain(“SIP/117-09afa988”, “”) in new stack
– <SIP/117-09afa988> Playing ‘vm-login’ (language ‘es’)
– <SIP/117-09afa988> Playing ‘vm-password’ (language ‘es’)
– <SIP/117-09afa988> Playing ‘vm-youhave’ (language ‘es’)
[Oct 28 08:46:22] WARNING[5313]: file.c:563 ast_openstream_full: File digits/1M does not exist in any format
[Oct 28 08:46:22] WARNING[5313]: file.c:866 ast_streamfile: Unable to open digits/1M (format 0x4 (ulaw)): No such file or directory
== Spawn extension (hotel, 8000, 1) exited non-zero on ‘SIP/117-09afa988’

ideas ?

thank forums …

ricky :cry:

2

Probably there is not a sound file in proper directory.

regards
darek

3

check what exactly you have in directory
/var/lib/asterisk/sounds/digits/
you miss file 1M.{some sound tuype - like gsm, wav etc}

4

I have my file 1M in the route /var/lib/asterisk/sounds/es/digits , not because asterisk does not read it, I have two files 1M.gsm and the other that rename to 1M

if change my language to us asterisk does not have problems to read my voicemail

<SIP/130-0897d3b0> Playing ‘vm-login’ (language ‘es’)
– <SIP/130-0897d3b0> Playing ‘vm-password’ (language ‘es’)
[Jan 3 03:33:17] WARNING[7508]: app_voicemail.c:6288 vm_authenticate: Unable to read password
– Executing [8000@hotel:1] VoiceMailMain(“SIP/130-089851c8”, “”) in new stack
– <SIP/130-089851c8> Playing ‘vm-login’ (language ‘es’)
– <SIP/130-089851c8> Playing ‘vm-password’ (language ‘es’)
– <SIP/130-089851c8> Playing ‘vm-youhave’ (language ‘es’)
[Jan 3 03:33:32] WARNING[7512]: file.c:563 ast_openstream_full: File digits/1M does not exist in any format
:confused:

5

Set the debugging and verbose high and see where asterisk is trying to play the file from.

6

— (0 headers 1 lines) —
[Dec 31 19:42:33] WARNING[6766]: file.c:563 ast_openstream_full: File digits/1M does not exist in any format
[Dec 31 19:42:33] WARNING[6766]: file.c:866 ast_streamfile: Unable to open digits/1M (format 0x4 (ulaw)): No such file or directory
== Spawn extension (hotel, 8000, 1) exited non-zero on 'SIP/130-0824ef00’
Scheduling destruction of SIP dialog ‘2631b8fe-348795bd@197.0.0.205’ in 32000 ms (Method: ACK)
set_destination: Parsing sip:130@2.2.2.2:49691 for address/port to send to
set_destination: set destination to 2.2.2.2, port 49691
Reliably Transmitting (NAT) to 2.2.2.2:49691:
BYE sip:130@2.2.2.2:49691 SIP/2.0
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4b866ea9;rport
From: “Voice Mail” sip:8000@1.1.1.1;tag=as52c5db25
To: sip:130@1.1.1.1;tag=a5d1c80af359f221o1
Call-ID: 2631b8fe-348795bd@197.0.0.205
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

7

Asterisk seems to want to play the file in .ulaw. If you do ls *.wav, ls *.ulaw in /usr/lib/asterisk/sounds what do you get ? Also you may want to install the asterisk sounds package (if you have not already done so).

8

I have file with extension .gsm

0.gsm 20.gsm 80.gsm day-6.gsm h-1.gsm million.gsm mon-8.gsm
10.gsm 2.gsm 8.gsm

within the path

/var/lib/asterisk/sounds/es/digits

my predefined language is in Spanish, but if I put in ingles I do not have problems

9

The problem is that you have no files in ulaw format. From what I understand asterisk should do the translation as long as you have the option available. You can try to convert the files to ulaw using sox.
sox.sourceforge.net/Main/HomePage
linux.die.net/man/1/sox
(I forgot the exact command but you can convert all the files over at once from gsm to ulaw)
You can also use the convert module in asterisk:
redice.krisk.org/