Voicemail not finding wav files

Beta Program Issues
1

Hi All,

Hopefully someone can help me out here. I’ve just upgraded to FreePBX 2.6.0.beta2.0 and since the upgrade, voicemail isn’t working correctly.

I can dial in to listen to messages etc but if a call goes to voicemail it just drops the call.

The log file shows the following.

[Aug 28 15:47:31] DEBUG[3439] app_macro.c: Executed application: Answer
[Aug 28 15:47:31] VERBOSE[3439] logger.c: – Executing [vmx@macro-vm:18] Read(“SIP/201-08416920”, “ACTION|/var/spool/asterisk/voicemail/default/212/unavail|1|skip|1|2”) in new stack
[Aug 28 15:47:31] VERBOSE[3439] logger.c: – Accepting a maximum of 1 digits.
[Aug 28 15:47:31] WARNING[3439] file.c: File /var/spool/asterisk/voicemail/default/212/unavail does not exist in any format
[Aug 28 15:47:31] WARNING[3439] file.c: Unable to open /var/spool/asterisk/voicemail/default/212/unavail (format 0x4 (ulaw)): No such file or directory
[Aug 28 15:47:31] VERBOSE[3439] logger.c: – User disconnected
[Aug 28 15:47:31] VERBOSE[3439] logger.c: == Spawn extension (macro-vm, vmx, 18) exited non-zero on ‘SIP/201-08416920’ in macro ‘vm’

I’ve looked in /var/spool/asterisk/voicemail/default/212/ and there is an unavail directory (which is empty as there hasn’t been a recorded custom message) and I’m not sure if this is correct.

I have just tried a user that has a recorded custom message and that just palys the default message…but is at least working.

Any advice would be very gratefully received.

Thanks.

Dave.

2

it does not sound like something that should be related t the upgrade. What version of Asterisk are you running?

3

Hi Philippe,

Thanks for your reply.

I’m running 1.4.16.2

Thanks for any help you can provide.

Regards,

Dave.

4

Do you have the Asterisk SIP Settings module installed? If you have that, please do a Check for updates online and update all modules that have updates.

Check your sip_general_additional.conf and verify that you have at least the follwoing:

disallow=all
allow=ulaw
allow=alaw
allow=gsm
5

Hi,

I do have this module installed, I have just checked for updates and everything is current.

In the sip_general_additional.conf file (in /etc/asterisk) I do have these options, below is a copy.

Thanks again for your help with this.

Regards,

Dave.

;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ;
; this file must be done via the web gui. There are alternative files to make ;
; custom modifications, details at: http://freepbx.org/configuration_files ;
;--------------------------------------------------------------------------------;
;

vmexten=*97
context=from-sip-external
callerid=Unknown
notifyringing=yes
notifyhold=yes
limitonpeers=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
disallow=all
allow=ulaw
allow=alaw
allow=gsm