Voicemail troubles

vinodh_css
Joined: Mon 18 of Jan, 2010

Re: Voicemail troubles

Posted:Mon 17 of May, 2010 (06:01 UTC)
Hi,

I see that you have set autoload=no in modules.conf.

In this case you have to load each and every application individually. If you set autoload=yes, all the application will be loaded by default.

Regards,
Vinodh
sleepymatt
Joined: Wed 12 of May, 2010

Re: Voicemail troubles

Posted:Fri 14 of May, 2010 (07:46 UTC)
Awesome. Thanks Vinod!

It was actually that the voicemail module was not being loaded. I simply added load => app_voicemail.so to modules.conf, restarted and bingo all is good!

Thanks again for the help!
vinodh_css
Joined: Mon 18 of Jan, 2010

Re: Voicemail troubles

Posted:Fri 14 of May, 2010 (05:23 UTC)
Hi,

I am not sure about what you have updated in modules.conf. But find simple configuration I used which worked without any issues.


SIP.conf
-----------
[5001]
type=friend
context=Main
host=dynamic
secret=1234
mailbox=5001@VM
pickupgroup=1
callgroup=1


Voicemail.conf
-----------------
[VM]
5000 = 1234,vinodh
5001 = 1234,vinodh


extensions.conf
---------------------
[Main]
include => voicemailmain
include => extns

[extns]
exten = 5001,1,Dial(SIP/5001,15)
exten = 5001,2,voicemail(5001@VM)
exten = 5001,3,Hangup()

exten = 5000,1,Dial(SIP/5000,15)
exten = 5000,2,voicemail(5000@VM)
exten = 5000,3,Hangup()

[voicemailmain]
exten = 5050,1,Answer()
exten = 5050,2,voicemailMain()
exten = 5050,3,Hangup()


In sip.conf User are configured with Main context. In extensions.conf the extensions are created under extns and voicemailmain configured separately. But both extns and voicemailmain are included in Main context.

Hope this gives some ideas.

Regards,
Vinodh
sleepymatt
Joined: Wed 12 of May, 2010

Re: Voicemail troubles

Posted:Thu 13 of May, 2010 (07:48 UTC)
Hey Vinod,

Thanks for replying!

I changed the extension 9999 in extensions.conf to exten => 9999,1,VoicemailMain() and refreshed/restarted but I am still getting the same thing in the cli:
== Using SIP RTP CoS mark 5
== Spawn extension (devices, 9999, 1) exited non-zero on 'SIP/200-00000000'

I don't understand what you mean by mapping the Voicemailmain context in sip.conf?

For the 200 user I have the following in sip.conf:
[200]
deny=0.0.0.0/0.0.0.0
callerid=************
type=friend
secret=secret
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=yes
mailbox=200@default
host=dynamic
dtmfmode=rfc2833
dial=SIP/200
context=devices
canreinvite=no
callgroup=
callerid=device <200>
accountcode=
call-limit=50


I don't know if this is possible but the only thing I can think of that could be causing this is that in the artice I followed to install asterisk said that we would be using our providers voicemail. I also was told to paste the following into modules.conf so maybe there is a voicemail module that is not being loaded??
[modules]
autoload=no                    ; Only load explicitely declared modules
load => format_pcm.so          ; Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G.711
load => codec_ulaw.so          ; mu-Law Coder/Decoder
load => app_dial.so            ; Dialing Application
load => app_macro.so           ; Extension Macros
load => app_playback.so        ; Sound File Playback Application
load => app_mixmonitor.so      ; Record calls
load => app_setcallerid.so     ; Set CallerID Presentation Application
load => app_disa.so            ; DISA - used for calling card-type projects
load => app_transfer.so        ; Transfer calls - good for use with DISA so that you don't proxy audio
load => func_timeout.so        ; Adjust timeout; handy for use with DISA.  But not essential if you can dial quickly.
load => func_callerid.so       ; Caller ID related dialplan functions
load => func_logic.so          ; GotoIf() and friends
load => func_strings.so        ; String handling functions
load => pbx_config.so          ; This loads your dialplan
load => pbx_spool.so           ; This is needed to make call files work
load => chan_sip.so            ; SIP
load => res_musiconhold.so     ; Music-on-Hold
load => func_shell.so          ; Execute shell commands and use the output in the dialplan.  (Useful for formatting things with gnu-sed.)
load => func_channel.so        ; Find information about the channel.  (Used with our implementation of DISA.)
 
; We do not use the following modules but mention them because they are common.
;load => format_wav.so          ; Microsoft WAV format
;load => app_echo.so            ; Simple Echo Application 
;load => res_features.so        ; Call parking



Thanks for your time, Vinod :)
vinodh_css
Joined: Mon 18 of Jan, 2010

Re: Voicemail troubles

Posted:Thu 13 of May, 2010 (03:50 UTC)
Hi,

Can you try VoicemailMain with empty brakets. It should ask for the mailbox number and the password.

exten => 9999,1,VoicemailMain()

check what is the context used in sip.conf for users and make sure you have this VoicemailMain context mapped to it. Also make sure you do a <reload> and <restart now> commands in asterisk console...

Regards,
Vinodh
sleepymatt
Joined: Wed 12 of May, 2010

Voicemail troubles

Posted:Wed 12 of May, 2010 (06:27 UTC)
Hey Guys,

I am new to asterisk. I have installed it on a Asus WL520GU Router and after spending the last 2 days finally have it making and receiving calls. Anyways, of all the things to setup Voicemail seems to be the easiest but it just wont work! I am running 1.6.

This is in my voicemail.conf
[general]
format=wav49|gsm|wav
attach=yes
[default]
200 => 1234,Matt,me@me.com

relevant part of extensions.conf
exten => 9999,1,VoiceMailMain(200@default)

exten => 9991,1,Voicemail(200@default)
exten => 9991,2,Hangup()

THe CLI outputs this:
  == Using SIP RTP CoS mark 5
  == Spawn extension (devices, 9999, 1) exited non-zero on 'SIP/200-00000019'
unknown*CLI>
  == Using SIP RTP CoS mark 5
  == Spawn extension (devices, 9991, 1) exited non-zero on 'SIP/200-0000001a'


I just don't get it. Help me!

Thanks very much!