deutsche voicemail klappt nicht

[iponer]

hallo möchte unter trixbox 2.8.0.3 einen deutschen anrufbeantworter haben.
habe hierfür von http://www.greenable.de/index.php?web=asterisk das paket
"core-prompts-DE-greenable.tar.gz" heruntergeladen und entpackt, die ganzen dateien liegen nun in "/var/lib/asterisk/sounds/de"

auf der trixbox-oberfläche habe ich dann bei einer extension den language code auf "de" gesetzt.

die ansagen kommen aber noch in englisch, was mache ich falsch?

gruss iponer

 

[woprr]

Bei verbose=5 werden die Dateizugriffsfehlermeldungen in der CLI angezeigt während der Ansage.
Vermutlich ist die Directory-Organisation in dem Paket an die von * anzupassen.

 

[iponer]

ja der spielt mir die dateien in "/var/lib/asterisk/sounds/en"
ab und nicht in "/var/lib/asterisk/sounds/de"
sorry ich bin jetzt kein grosser linux-experte, wo genau stell ich das in den konfigurationsfiles um? (oder in der weboberfläche)

 

[woprr]

asterisk.conf

 

[iponer]

[directories](!) ; remove the (!) to enable this
astetcdir => /etc/asterisk
astmoddir => /usr/lib/asterisk/modules
astvarlibdir => /var/lib/asterisk
astdbdir => /var/lib/asterisk
astkeydir => /var/lib/asterisk
astdatadir => /var/lib/asterisk
astagidir => /var/lib/asterisk/agi-bin
astspooldir => /var/spool/asterisk
astrundir => /var/run/asterisk
astlogdir => /var/log/asterisk

;[options]
;verbose = 3
;debug = 3
;alwaysfork = yes ; same as -F at startup
;nofork = yes ; same as -f at startup
;quiet = yes ; same as -q at startup
;timestamp = yes ; same as -T at startup
;execincludes = yes ; support #exec in config files
;console = yes ; Run as console (same as -c at startup)
;highpriority = yes ; Run realtime priority (same as -p at startup)
;initcrypto = yes ; Initialize crypto keys (same as -i at startup)
;nocolor = yes ; Disable console colors
;dontwarn = yes ; Disable some warnings
;dumpcore = yes ; Dump core on crash (same as -g at startup)
;languageprefix = yes ; Use the new sound prefix path syntax
;internal_timing = yes
;systemname = my_system_name ; prefix uniqueid with a system name for global uniqueness issues
;autosystemname = yes ; automatically set systemname to hostname - uses 'localhost' on failure, or systemname if set
;maxcalls = 10 ; Maximum amount of calls allowed
;maxload = 0.9 ; Asterisk stops accepting new calls if the load average exceed this limit
;maxfiles = 1000 ; Maximum amount of openfiles
;minmemfree = 1 ; in MBs, Asterisk stops accepting new calls if the amount of free memory falls below this watermark
;cache_record_files = yes ; Cache recorded sound files to another directory during recording
;record_cache_dir = /tmp ; Specify cache directory (used in cnjunction with cache_record_files)
;transmit_silence_during_record = yes ; Transmit SLINEAR silence while a channel is being recorded
;transmit_silence = yes ; Transmit SLINEAR silence while a channel is being recorded or DTMF is being generated
;transcode_via_sln = yes ; Build transcode paths via SLINEAR, instead of directly
;runuser = asterisk ; The user to run as
;rungroup = asterisk ; The group to run as

; Changing the following lines may compromise your security.
;[files]
;astctlpermissions = 0660
;astctlowner = root
;astctlgroup = apache
;astctl = asterisk.ctl

[compat]
pbx_realtime=1.6
res_agi=1.6
app_set=1.6
language=de

wo soll ich das umstellen und wie?
language=de hab ich noch dazugeschrieben

 

[woprr]

+ astsoundsdir => /usr/share/asterisk/sounds(/de/...)
?

 

[iponer]

[directories](!) ; remove the (!) to enable this
astetcdir => /etc/asterisk
astmoddir => /usr/lib/asterisk/modules
astvarlibdir => /var/lib/asterisk
astdbdir => /var/lib/asterisk
astkeydir => /var/lib/asterisk
astdatadir => /var/lib/asterisk
astagidir => /var/lib/asterisk/agi-bin
astspooldir => /var/spool/asterisk
astrundir => /var/run/asterisk
astlogdir => /var/log/asterisk
astsoundsdir => /usr/share/asterisk/sounds/de

;[options]
;verbose = 3
;debug = 3
;alwaysfork = yes ; same as -F at startup
;nofork = yes ; same as -f at startup
;quiet = yes ; same as -q at startup
;timestamp = yes ; same as -T at startup
;execincludes = yes ; support #exec in config files
;console = yes ; Run as console (same as -c at startup)
;highpriority = yes ; Run realtime priority (same as -p at startup)
;initcrypto = yes ; Initialize crypto keys (same as -i at startup)
;nocolor = yes ; Disable console colors
;dontwarn = yes ; Disable some warnings
;dumpcore = yes ; Dump core on crash (same as -g at startup)
;languageprefix = yes ; Use the new sound prefix path syntax
;internal_timing = yes
;systemname = my_system_name ; prefix uniqueid with a system name for global uniqueness issues
;autosystemname = yes ; automatically set systemname to hostname - uses 'localhost' on failure, or systemname if set
;maxcalls = 10 ; Maximum amount of calls allowed
;maxload = 0.9 ; Asterisk stops accepting new calls if the load average exceed this limit
;maxfiles = 1000 ; Maximum amount of openfiles
;minmemfree = 1 ; in MBs, Asterisk stops accepting new calls if the amount of free memory falls below this watermark
;cache_record_files = yes ; Cache recorded sound files to another directory during recording
;record_cache_dir = /tmp ; Specify cache directory (used in cnjunction with cache_record_files)
;transmit_silence_during_record = yes ; Transmit SLINEAR silence while a channel is being recorded
;transmit_silence = yes ; Transmit SLINEAR silence while a channel is being recorded or DTMF is being generated
;transcode_via_sln = yes ; Build transcode paths via SLINEAR, instead of directly
;runuser = asterisk ; The user to run as
;rungroup = asterisk ; The group to run as

; Changing the following lines may compromise your security.
;[files]
;astctlpermissions = 0660
;astctlowner = root
;astctlgroup = apache
;astctl = asterisk.ctl

[compat]
pbx_realtime=1.6
res_agi=1.6
app_set=1.6
language=de

hmm geht irgendwie noch nicht

 

[rentier-s]

Damit Asterisk die Verzeichniseinträge beachtet, musst Du das (!) in der ersten Zeile entfernen.
"remove the (!) to enable this"

Die language kann man bei den meisten Channel-Treibern in der jeweiligen config-Datei einstellen, zB. sip.conf. Bei der Trixbox musst Du evtl. eine sip_custom.conf anlegen und diese in der sip.conf includen, damit die Einstellungen nicht vom GUI überschrieben werden.

Außerdem musst Du bei neueren Asterisk-Versionen auf korrekte Verzeichnisstruktur passend zu Deiner languageprefix Einstellung achten, damit digits und letters gefunden werden.

Zur Not hilft im Dialplan ein Set(CHANNEL(language)=de), um die Sprache des jeweiligen Anrufs zu setzen. Frag mich aber nicht, wie man das im GUI unterbringt.

Rentier

 

[iponer]

so hab den alten rotz deinstalliert und das ganze mithilfe dieser anleitung realisiert:

http://www.linux-fuer-blinde.de/176-0-deutsche-sprachbausteine-fuer-asterisk-installieren.html

hat wunderbar einfach funktioniert

 

[iponer]

ok dafür ist jetzt eine andere sache aufgetaucht:

hab 2 trunks: einmal für sipgate basic (prvt nummer) und sipgate team edition/trunking (buisness nummern)
die beiden benutzen eine inbound-route die auf eine extension verweist
wenn ich jetzt auf sipgate basic anrufe kommt die deutsche voicemail, wenn ich auf eine nummer anrufe die bei sipgate team edition hinterlegt ist bekomm ich die englische voicemail-ansage... ich möchte aber in beiden fällen die deutsche haben.

asterisk.conf sieht so aus:

[directories]
astetcdir => /etc/asterisk
astmoddir => /usr/lib/asterisk/modules
astvarlibdir => /var/lib/asterisk
astdbdir => /var/lib/asterisk
astkeydir => /var/lib/asterisk
astdatadir => /var/lib/asterisk
astagidir => /var/lib/asterisk/agi-bin
astspooldir => /var/spool/asterisk
astrundir => /var/run/asterisk
astlogdir => /var/log/asterisk
astsoundsdir => /var/lib/asterisk/sounds/de

;[options]
;verbose = 3
;debug = 3
;alwaysfork = yes ; same as -F at startup
;nofork = yes ; same as -f at startup
;quiet = yes ; same as -q at startup
;timestamp = yes ; same as -T at startup
;execincludes = yes ; support #exec in config files
;console = yes ; Run as console (same as -c at startup)
;highpriority = yes ; Run realtime priority (same as -p at startup)
;initcrypto = yes ; Initialize crypto keys (same as -i at startup)
;nocolor = yes ; Disable console colors
;dontwarn = yes ; Disable some warnings
;dumpcore = yes ; Dump core on crash (same as -g at startup)
;languageprefix = yes ; Use the new sound prefix path syntax
;internal_timing = yes
;systemname = my_system_name ; prefix uniqueid with a system name for global uniqueness issues
;autosystemname = yes ; automatically set systemname to hostname - uses 'localhost' on failure, or systemname if set
;maxcalls = 10 ; Maximum amount of calls allowed
;maxload = 0.9 ; Asterisk stops accepting new calls if the load average exceed this limit
;maxfiles = 1000 ; Maximum amount of openfiles
;minmemfree = 1 ; in MBs, Asterisk stops accepting new calls if the amount of free memory falls below this watermark
;cache_record_files = yes ; Cache recorded sound files to another directory during recording
;record_cache_dir = /tmp ; Specify cache directory (used in cnjunction with cache_record_files)
;transmit_silence_during_record = yes ; Transmit SLINEAR silence while a channel is being recorded
;transmit_silence = yes ; Transmit SLINEAR silence while a channel is being recorded or DTMF is being generated
;transcode_via_sln = yes ; Build transcode paths via SLINEAR, instead of directly
;runuser = asterisk ; The user to run as
;rungroup = asterisk ; The group to run as

; Changing the following lines may compromise your security.
;[files]
;astctlpermissions = 0660
;astctlowner = root
;astctlgroup = apache
;astctl = asterisk.ctl

[compat]
pbx_realtime=1.6
res_agi=1.6
app_set=1.6
languageprefix=yes

hab schin vieles durchprobiert, aber noch ohne erfolg :/