[Asterisk-Users] Problem locating stream files

[Daniel Gonzalez]

Hi *,

I have set up a very simple asterisk configuration where I intend to be redirected to the
voicemail whenever I dial 100 with my kphone SIP client. The problem is that asterisk can not find
the stream 'vm-theperson'. I have made a non-standard installation (since I am just testing), and
that file is located in /mnt/tr2/fake_root/installed/usr/local/var/lib/asterisk/sounds.

1. How can I tell asterisk where to look for the streams? 
2. I have traced (with strace) all 6 threads created by asterisk to find out which directories it
is accessing to try to open the vm-theperson stream, but interestingly this system call does not
show at all in the traces. How is this possible? My understanding is that any system call
initiated by the process (be it in the core process, or in any shared library) should show up in
strace, but I am not that proficient with strace.


Here you have some details. First, the asterisk traces:

    -- Registered SIP 'gonvaled' at 192.168.1.200 port 5060 expires 900
    -- Executing VoiceMail("SIP/gonvaled-e0c6", "u100") in new stack
Jul  2 11:53:32 WARNING[81926]: file.c:462 ast_openstream: File vm-theperson does not exist in any
format
Jul  2 11:53:32 WARNING[81926]: file.c:750 ast_streamfile: Unable to open vm-theperson (format
ULAW): File exists
  == Spawn extension (from-sip, 100, 1) exited non-zero on 'SIP/gonvaled-e0c6'


This is my extensions.conf:

[from-sip]
exten => 100,1,Voicemail(u100)
exten => 100,2,Hangup

exten => 999,1,VoicemailMain(${CALLERIDNUM})

---------------------------------------------------- End of extensions.conf


This is my sip.conf

[general]
port = 5060                     ; Port to bind to
bindaddr = 0.0.0.0              ; Address to bind SIP channel to
context = default               ; Default context for incoming calls
disallow=all                    ; Disallow all codecs
allow=ulaw                      ; Allow codecs in order of preference

[gonvaled]
type=friend
host=dynamic
dtmfmode=inband                 ; Choices are inband, rfc2833, or info
username=gonvaled
secret=blah
context=from-sip
mailbox=100

---------------------------------------------------- End of sip.conf

And this is my modules.conf

;
; Module Loader configuration file
;

[modules]
;autoload=yes
;
; If you want, load the GTK console right away.
; Don't load the KDE console since
; it's not as sophisticated right now.
;
load => pbx_gtkconsole.so
;load => pbx_gtkconsole.so
noload => pbx_kdeconsole.so
;
; Intercom application is obsoleted by
; chan_oss.  Don't load it.
;
noload => app_intercom.so
;
; Explicitly load the chan_modem.so early on to be sure
; it loads before any of the chan_modem_* 's afte rit
;
;load => chan_modem.so
load => res_musiconhold.so

;load => app_dial.so

; Needed for CAPI
load => chan_capi.so

; Needed for SIP
load => res_parking.so
load => chan_sip.so

load => pbx_config.so
;
; Load either OSS or ALSA, not both
; By default, load OSS only (automatically) and do not load ALSA
;
noload => chan_alsa.so
;noload => chan_oss.so

; Needed for voicemail
load => res_adsi.so
load => app_voicemail.so
;
; Module names listed in "global" section will have symbols globally
; exported to modules loaded after them.
;
[global]
;chan_modem.so=yes
chan_capi.so=yes




Thanks for your help,

Daniel Gonzalez