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Spoke problems with the Asterisk


2010-09-07
 
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Hello,
I am occupied now since a few days with the Asterisk.
It runs here 1.4.20 on a current Debian.

However I have problems with the acoustic output of the mailbox.
I have the language packages of http://www.ip-phone-forum.de/showthread.php/www.amooma.de brought in how
it there is described. Afterwards I have into the Sip.conf the language
changed over to “de”. But it changed nothing in the language.
But there is now another problem. If I on the mailbox of one mine
both telephones to speak (the numbers are 10 and 11) breaks those would like
Announcement simply starting from and in the log stands which it the sound file 1f.gsm
or 1-f.gsm not to find can.
There I did not overwrite the language files and it before problem-free
functioned surprises me some little.

P.S. I start the Asterisk again after each change in the Config files…
with ./asterisk restart in init.d

Thanks in advance
Debug in the CLI stop and the expenditure here posts.
Sorry lasted it somewhat longer.
Here the expenditure of the Asterisk with Bose = 10 and debug = 10

I called the mailbox inquiry (90) from branch 10.
There came as English message “she has” and then still broke off he.
If I on English change go it over problem-free.

Here the expenditure of the CONSOLE
Code: 
<----------->
    -- Executing [90@default: 2] VoiceMailMain (“SIP/10-081d3e88”, “10”) in new stack
    --  <SIP/10-081d3e88> Playing “vm-password” (LANGUAGE “de”)
 inux*CLI>
<-- SIP READ from 192.168.0.10: 6988 -->
ACK sip: 90@ 192.168.0.2 SIP/2.0
 via: SIP/2.0/UDP 192.168.0.10: 6988; branch=z9hG4bK-d8754z-e6610b731b054e21-1--d8754z; rport
 max Forwards: 70
 Contact: <sip: 10@ 192.168.0.10: 6988>
Ton: “90”< sip: 90@ 192.168.0.2>; tag=as48342636
 From: “Asterisk”< sip: 10@ 192.168.0.2>; tag=7f2b8b75
 call ID: Mjc1N2RjYWRhM2RlYjY5ODg5OGNhNTE4NzE3ZmJhMzk.
CSeq: 2 ACK
 Proxy Authorization: Digest username= " 10 ", realm= " asterisk ", nonce= " 0f5461b9 ", uri= " sip: 90@ 192.168.0.2 ", response= " 093b342f3b10e65cc7b619a15afc3f1a ", algorithm=MD5
 user agent: X-Lite release 1100l stamp 47546
 content length: 0


<------------>
-- (11 header 0 LINEs) --
  inux*CLI>
<-- SIP READ from 192.168.0.10: 6988 -->



<------------>
Really destroying SIP dialogue “YzdmMmM5MmY3NDI0NGU0OTJjZDZjMjAyZjY2ZDg3NDY.” Method: REGISTER
    --  <SIP/10-081d3e88> Playing “vm-youhave” (LANGUAGE “de”)
 [July 20 12:15: 54] WARNING [16557]: file.c: 602 ast_openstream_full: File digits/1F of does emergency exist in any format
 [July 20 12:15: 54] WARNING [16557]: file.c: 912 ast_streamfile: Unable ton of open digits/1F (format 0x4 (ulaw)): NO look for file or directory
  == Spawn extension (default, 90, 2) exited non zero on “SIP/10-081d3e88”
 Scheduling destruction OF SIP dialogue “Mjc1N2RjYWRhM2RlYjY5ODg5OGNhNTE4NzE3ZmJhMzk.” in 32000 ms (Method: ACK)
 set_destination: Parsing <sip: 10@ 192.168.0.10: 6988> for to ADDRESS/haven ton send tons
 set_destination: set destination ton of 192.168.0.10, haven 6988
 Reliably Transmitting (NO NAT) ton of 192.168.0.10: 6988:
BYE sip: 10@ 192.168.0.10: 6988 SIP/2.0
 via: SIP/2.0/UDP 192.168.0.2: 5060; branch=z9hG4bK19239aba; rport
 From: “90”< sip: 90@ 192.168.0.2>; tag=as48342636
 ton: “Asterisk”< sip: 10@ 192.168.0.2>; tag=7f2b8b75
 call ID: Mjc1N2RjYWRhM2RlYjY5ODg5OGNhNTE4NzE3ZmJhMzk.
CSeq: 102 BYE
 user agent: Asterisk PBX
 max Forwards: 70
 content length: 0
Quotation:
[July 20 12:15: 54] WARNING [16557]: file.c: 602 ast_openstream_full: File digits/1F of does emergency exist in any format
[July 20 12:15: 54] WARNING [16557]: file.c: 912 ast_streamfile: Unable ton of open digits/1F (format 0x4 (ulaw)): NO look for file or directory
, There you made probably TC somewhat wrong with the installation of the sound files.
The sound files are installed after guidance.
Which surprises me much more is that the announcement in English is
and it with switch to the English language problem-free functioned
but with the German language not although it is spent in English.
the problem installed I yesterday also, likewise after guidance and went nich.
I think times is connected with that installation kind, has * with apt GET installed, there it otherwise problme with headers on the ip1101 gives.
solution was rather simple.
Code: 
apt GET install asterisk prompt de
and stop into sip.conf the language on DE change, and now lufts
I will times try that out afterwards.
If should be so simple…
It goes… and without problems.
Super… thank you. There I could myself have looked for still black.
Quotation:
Quotation of Arcon Contribution indicate the problem installed I yesterday also, likewise after guidance and went nich.
I think times is connected with that installation kind, has * with apt GET installed, there it otherwise problme with headers on the ip1101 gives.
solution was rather simple.
Code: 
apt GET install asterisk prompt de
and stop into sip.conf the language on DE change, and now lufts
Afterwards now my Asterisk does not start any longer.
with which error does it break start off?
Quotation:
Quotation of Arcon Contribution indicate with which error does it break start off?
No error comes. Naja I white also why, apt GET has nen Asterisk installed.
with apt GET always read which it to make wants!

schmeiss new simply again by apt GET remove down.

if you installed your “old” asterisk manuel, must also the language packages in such a way install
Code: 
[Oct 24 10:32: 06] WARNING [27921]: file.c: 582 ast_openstream_full: File digits/1F of does emergency exist in any format
 [Oct 24 10:32: 06] WARNING [27921]: file.c: 891 ast_streamfile: Unable ton of open digits/1F (format 0x4 (ulaw)): NO look for file or directory
With me ises nearly just as only that also on German is and at the beginning also functioned nevertheless then to comb sone Fehlermaeldung if I English switches goes to ' s again.
that pack asterisk prompt de hangs of pack asterisk off. That should actually also communicate “apt GET” to you. During the guidance of amooma.de I can recommend the “old person version” (for Asterisk 1,2 and 1,4) from Gabi Becker.
(Here is the drawing eating jerk door however different… see below)

Here is the list of the files pack from the Debian etch. According to amooma.de guidance lie all into “… /asterisk/sounds/de”. In addition one should then also 'languageprefix=yes'set in “asterisk.conf”.

Here still another reason for the “digits/1F”: In that app_voicemail.c a remark stands:
I admit, rather hidden… Code: 
German of requires the following additional sound file:
1F	(feminine)
Here in the forum there were also times one Contribution.
Also for the 1.6er does version function and as can one the before installed Packete again deinstallieren? Which I must make it again go I am still quite unerfahern which Asterisk concern.
Thus the problem is not really gelsst probably or?

Under http://http://www.ip-phone-forum.de/showthread.php/www.amooma.de/asterisk/sprachbausteine/ there are sound files
for Asterisk 1,2 and 1.4 -> http://http://www.ip-phone-forum.de/showthread.php/www.amooma.de/asterisk/sprach...prompts.tar.gz
and Asterisk 1,4 and 1.6 -> http://http://www.ip-phone-forum.de/showthread.php/www.amooma.de/asterisk/sprach...0080705.tar.gz

Strangely enough there that stands the old version to be incomplete is. But this contains a 1F. In the current version (with another voice: -/) this 1F is missing. A umkopieren one can make, sounds themselves however amusingly.
But the problem was that I had to rename the file 1 in 1F go now everything problem-free. But thanks again

I scheib again ne guidance if different the same problem have
Aso the Prblem with the language components came after 1 week again and also after one backup ises still remained in addition me meanwhile many more warnings was angezeiget.

Code: 
Asterisk 1.4.21-BRIstuffed-0.4.0-RC3, copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Marks of Spencer <markster@digium.com>
Asterisk of comes with ABSOLUTELY NO WARRANTY; type “core show warranty” for details.
This is free software, with components licensed more under the GNU general publicly
 License version 2 and OTHER of licenses; you acres welcome tons redistribute it more under
 certain conditions. Type “core show license” for details.
=========================================================================
 Connected ton of Asterisk 1.4.21-BRIstuffed-0.4.0-RC3 currently running on srvas011 (pid = 3278)
 srvas011*CLI> reload
 The “reload” COMMANDs is deprecated and wants removed in A the future releases. Please use “modules reload” instead.
[DEK 18 13:22: 32] NOTICE [3311]: cdr.c: 1373 do_reload: CDR simple LOGGING enabled.
[DEK 18 13:22: 32] WARNING [3311]: res_musiconhold.c: 1068 load_moh_classes: A directory must specified for class “general”!
[DEK 18 13:22: 32] NOTICE [3311]: indications.c: 505 ast_unregister_indication_country: Removed default indication country “de”
 [DEK 18 13:22: 32] WARNING [3311]: res_features.c: 3055 load_config:  is emergency A valid parkingtime
 [DEK 18 13:22: 32] ERROR [3311]: func_odbc.c: 621 reload: CAN emergency initialize query SQL
 [DEK 18 13:22: 32] ERROR [3311]: func_odbc.c: 621 reload: CAN emergency initialize query ANTIGF
 [DEK 18 13:22: 32] ERROR [3311]: func_odbc.c: 621 reload: CAN emergency initialize query PRESENCE
 [DEK 18 13:22: 32] NOTICE [3311]: app_playback.c: 457 reload: Reloading say.conf
 [DEK 18 13:22: 32] NOTICE [3311]: pbx_ael.c: 4131 pbx_load_module: Starting AEL load process.
[DEK 18 13:22: 32] NOTICE [3311]: pbx_ael.c: 4138 pbx_load_module: AEL load process: calculated config file name “/etc/asterisk/extensions.ael”.
[DEK 18 13:22: 32] NOTICE [3311]: pbx_ael.c: 4146 pbx_load_module: AEL load process: parsed config file name “/etc/asterisk/extensions.ael”.
[DEK 18 13:22: 32] NOTICE [3311]: pbx_ael.c: 4149 pbx_load_module: AEL load process: checked config file name “/etc/asterisk/extensions.ael”.
[DEK 18 13:22: 32] NOTICE [3311]: pbx_ael.c: 4151 pbx_load_module: AEL load process: compiled config file name “/etc/asterisk/extensions.ael”.
[DEK 18 13:22: 32] NOTICE [3311]: pbx_ael.c: 4154 pbx_load_module: AEL load process: merged config file name “/etc/asterisk/extensions.ael”.
[DEK 18 13:22: 32] NOTICE [3311]: pbx_ael.c: 4157 pbx_load_module: AEL load process: verified config file name “/etc/asterisk/extensions.ael”.
[DEK 18 13:22: 32] WARNING [3297]: config.c: 806 process_text_line: NO “=” (equal sign) in LINE of 16615 OF /etc/asterisk/sip.conf
 [DEK 18 13:22: 32] WARNING [3297]: chan_sip.c: 16547 handle_common_options: Unknown canreinvite mode '' on LINE 995
 [DEK 18 13:22: 32] WARNING [3297]: chan_sip.c: 16516 handle_common_options: Unknown dtmf mode '' on LINE 1000, using rfc2833
 [DEK 18 13:22: 32] WARNING [3297]: frame.c: 1320 ast_parse_allow_disallow: CAN emergency allow unknown format “undefined”
 [DEK 18 13:22: 32] WARNING [3297]: chan_sip.c: 16547 handle_common_options: Unknown canreinvite mode '' on LINE 995
 [DEK 18 13:22: 32] WARNING [3297]: chan_sip.c: 16516 handle_common_options: Unknown dtmf mode '' on LINE 1000, using rfc2833
 [DEK 18 13:22: 32] WARNING [3297]: chan_sip.c: 17145 more build_peer: '' is emergency A valid RTP hold time RK LINE 1027.  Using default.
[DEK 18 13:22: 32] WARNING [3297]: chan_sip.c: 17140 more build_peer: '' is emergency A valid RTP hold time RK LINE 1028.  Using default.
[DEK 18 13:22: 32] WARNING [3297]: frame.c: 1320 ast_parse_allow_disallow: CAN emergency allow unknown format “undefined”
and with asterisk - vvvvvc
Code: 
[DEK 18 13:24: 52] WARNING [3368]: loader.c: 670 load_resource: Modules “format_sln16.so” could emergency loaded.
    == registered ones custom function MUSICCLASS
 func_moh.so => (music on hold dial-flat function)
  == Parsing “/etc/asterisk/extensions.conf”: Found
  == Setting global variable “CONSOLE” ton of “CONSOLE/dsp”
  == Setting global variable “IAXINFO” ton “guest”
  == Setting global variable “DRUNKENNESS” ton of “DAHDI/G2”
  == Setting global variable “TRUNKMSD” ton of “1”
  == Setting global variable “FEATURES” tons ''
  == Setting global variable “DIAL OPTION” ton ''
  == Setting global variable “RING TIME” ton of “20”
  == Setting global variable “FOLLOWMEOPTIONS” tons ''
  == Setting global variable “PAGING_HEADER” ton of “Intercom”
  == Setting global variable “PAGING_TIMEOUT” tons of “60”
    -- Registered one extension context “dundi-e164-canonical”
    -- Registered one extension context “dundi-e164-customers”
    -- Registered one extension context “dundi-e164-via-pstn”
    -- Registered one extension context “dundi-e164-local”
    -- Including context “dundi-e164-canonical” in context “dundi-e164-local”
    -- Including context “dundi-e164-customers” in context “dundi-e164-local”
    -- Including context “dundi-e164-via-pstn” in context “dundi-e164-local”
    -- Registered one extension context “dundi-e164-switch”
    -- Including SWITCH “DUNDi/e164” in context “dundi-e164-switch”
    -- Registered one extension context “dundi-e164-lookup”
    -- Including context “dundi-e164-local” in context “dundi-e164-lookup”
    -- Including context “dundi-e164-switch” in context “dundi-e164-lookup”
    -- Registered one extension context “macro-dundi-e164”
    -- Added extension “s” priority 1 ton of macro-dundi-e164
    -- Including context “dundi-e164-lookup” in context “macro-dundi-e164”
    -- Registered one extension context “iaxtel700”
    -- Added extension “” priority 1 ton of iaxtel700
    -- Registered one extension context “more iaxprovider”
    -- Registered one extension context “trunkint”
    -- Added extension “_9011.” priority 1 ton trunkint
    -- Added extension “_9011.” priority 2 tons trunkint
    -- Registered one extension context “trunkld”
    -- Added extension “” priority 1 ton trunkld
    -- Added extension “” priority 2 tons trunkld
    -- Registered one extension context “drunkenness-local”
    -- ADD Found
 pbx_config.so => (Text Extension Configuration)
  == Parsing “/etc/asterisk/codecs.conf”: Found
    -- CODEC SPEEX: Setting quality ton of 3
    -- CODEC SPEEX: Setting Complexity ton of 2
    -- CODEC SPEEX: Perceptual Enhancement mode. [on]
    -- CODEC SPEEX: VAD mode. [on]
    -- CODEC SPEEX: VBR mode. [on]
    -- CODEC SPEEX: Disabling ABR
    -- CODEC SPEEX: Setting VBR quality ton of 4.000000
    -- CODEC SPEEX: DTX mode. [off]
    -- CODEC SPEEX: Preprocessing. [off]
    -- CODEC SPEEX: Preprocessor VAD. [off]
    -- CODEC SPEEX: Preprocessor AGC. [off]
    -- CODEC SPEEX: Setting preprocessor AGC level ton of 8000.000000
    -- CODEC SPEEX: Preprocessor Denoise. [off]
    -- CODEC SPEEX: Preprocessor the verb. [off]
    -- CODEC SPEEX: Setting preprocessor the verb Decay ton of 0.400000
    -- CODEC SPEEX: Setting preprocessor the verb level ton of 0,300000
  == registered ones translator “speextolin” from format speex tons slin, cost 5
  == registered ones translator “lintospeex” from format slin tons speex, cost 41
 codec_speex.so => (Speex Coder/decoder)
  == Parsing “/etc/asterisk/cdr_custom.conf”: Found
 cdr_custom.so => (Customizable Comma Separated VALUEs CDR baking)
  == registered one application “SpeechCreate”
  == registered ones application “SpeechLoadGrammar”
  == registered ones application “SpeechUnloadGrammar”
  == registered ones application “SpeechActivateGrammar”
  == registered ones application “SpeechDeactivateGrammar”
  == registered ones application “SpeechStart”
  == registered ones application “SpeechBackground”
  == registered ones application “SpeechDestroy”
  == registered ones application “SpeechProcessingSound”
  == registered ones custom function SPEECH
  == registered ones custom function SPEECH_SCORE
  == registered ones custom function SPEECH_TEXT
  == registered ones custom function SPEECH_GRAMMAR
  == registered ones custom function SPEECH_ENGINE
  == registered ones custom function SPEECH_RESULTS_TYPE
 app_speech_utils.so => (Dial plan Speech Applications)
  == registered ones application “ParkAndAnnounce”
 app_parkandannounce.so => (Call Parking and Announce Application)
  == registered one file format vox, extension (s) vox
 format_vox.so => (Dia. logic VOX (ADPCM) file format)
  == Parsing “/etc/asterisk/codecs.conf”: Found
    -- codec_zap: using towards Eric PLC
  == NO hardware transcoders found.
codec_zap.so => (Gene Eric Zaptel Transcoder codec translator)
  == Parsing “/etc/asterisk/dundi.conf”: Found
  == Using TOS bit 0
  == DUNDi ready and Listening on 0.0.0.0 haven 4520
  == registered ones custom function DUNDILOOKUP
 pbx_dundi.so => (Distributed universal NUMBER Discovery (DUNDi))
[DEK 18 13:24: 52] NOTICE [3368]: pbx_ael.c: 4131 pbx_load_module: Starting AEL load process.
[DEK 18 13:24: 52] NOTICE [3368]: pbx_ael.c: 4138 pbx_load_module: AEL load process: calculated config file name “/etc/asterisk/extensions.ael”.
[DEK 18 13:24: 52] NOTICE [3368]: pbx_ael.c: 4146 pbx_load_module: AEL load process: parsed config file name “/etc/asterisk/extensions.ael”.
[DEK 18 13:24: 52] NOTICE [3368]: pbx_ael.c: 4149 pbx_load_module: AEL load process: checked config file name “/etc/asterisk/extensions.ael”.
    == Setting global variable “CONSOLE” ton of “CONSOLE/dsp”
  == Setting global variable “IAXINFO” ton “guest”
  == Setting global variable “DRUNKENNESS” ton of “DAHDI/G2”
  == Setting global variable “TRUNKMSD” ton of “1”
    -- Registered one extension context “ael-dundi-e164-canonical”
    -- Registered one extension context “ael-dundi-e164-customers”
    -- Registered one extension context “ael-dundi-e164-via-pstn”
    -- Registered one extension context “ael-dundi-e164-local”
    -- Including context “ael-dundi-e164-canonical” in context “ael-dundi-e164-local”
    -- Including context “ael-dundi-e164-customers” in context “ael-dundi-e164-local”
    -- Including context “ael-dundi-e164-via-pstn” in context “ael-dundi-e164-local”
    -- Registered one extension context “ael-dundi-e164-switch”
    -- Including SWITCH “DUNDi/e164” in context “ael-dundi-e164-switch”
    -- Registered one extension context “ael-dundi-e164-lookup”
    -- Including context “ael-dundi-e164-local” in context “ael-dundi-e164-lookup”
    -- Including context “ael-dundi-e164-switch” in context “ael-dundi-e164-lookup”
    -- Registered one extension context “macro-ael-dundi-e164”
    -- Registered one extension context “ael-iaxtel700”
    -- Registered one extension context “ael more iaxprovider”
    -- Registered one extension context “ael trunkint”
    -- Including context “ael-dundi-e164-lookup” in context “ael trunkint”
    -- Registered one extension context “ael trunkld”
    -- Including context “ael-dundi-e164-lookup” in context “ael trunkld”
    -- Registered one extension context “ael trunklocal”
    -- Registered one extension context “ael trunktollfree”
    -- Registered one extension context “ael internationally”
    -- Including context “ael longdistance” in context “ael internationally”
    -- Including context “ael trunkint” in context “ael international”
    -- Registered one extension context “ael longdistance”
    -- Including context “ael local” in context “ael longdistance”
    -- Including context “ael trunkld” in context “ael longdistance”
    -- Registered one extension context “ael local”
    -- Includin pbx_ael.c: 4151 pbx_load_module: AEL load process: compiled config file name “/etc/asterisk/extensions.ael”.
[DEK 18 13:24: 52] NOTICE [3368]: pbx_ael.c: 4154 pbx_load_module: AEL load process: merged config file name “/etc/asterisk/extensions.ael”.
[DEK 18 13:24: 52] NOTICE [3368]: pbx_ael.c: 4157 pbx_load_module: AEL load process: verified config file name “/etc/asterisk/extensions.ael”.
pbx_ael.so => (Asterisk Extension LANGUAGE compiler)
  == registered ones application “ZapBarge”
 app_zapbarge.so => (Zap would save channel application in on)
  == Parsing “/etc/asterisk/sip.conf”: Found
 [DEK 18 13:24: 52] WARNING [3368]: config.c: 806 process_text_line: NO “=” (equal sign) in LINE of 16615 OF /etc/asterisk/sip.conf
  == Parsing “/etc/asterisk/users.conf”: Found
 [DEK 18 13:24: 52] WARNING [3368]: chan_sip.c: 16547 handle_common_options: Unknown canreinvite mode '' on LINE 995
 [DEK 18 13:24: 52] WARNING [3368]: chan_sip.c: 16516 handle_common_options: Unknown dtmf mode '' on LINE 1000, using rfc2833
 [DEK 18 13:24: 52] WARNING [3368]: frame.c: 1320 ast_parse_allow_disallow: CAN emergency allow unknown format “undefined”
 [DEK 18 13:24: 52] WARNING [3368]: chan_sip.c: 16547 handle_common_options: Unknown canreinvite mode '' on LINE 995
 [DEK 18 13:24: 52] WARNING [3368]: chan_sip.c: 16516 handle_common_options: Unknown dtmf mode '' on LINE 1000, using rfc2833
 [DEK 18 13:24: 52] WARNING [3368]: chan_sip.c: 17145 more build_peer: '' is emergency A valid RTP hold time RK LINE 1027.  Using default.
[DEK 18 13:24: 52] WARNING [3368]: chan_sip.c: 17140 more build_peer: '' is emergency A valid RTP hold time RK LINE 1028.  Using default.
[DEK 18 13:24: 52] WARNING [3368]: frame.c: 1320 ast_parse_allow_disallow: CAN emergency allow unknown format “undefined”
  == SIP Listening on 192.168.4.11: 5060
  == Using SIP TOS: none
  == Parsing “/etc/asterisk/sip_notify.conf”: Found
  == registered ones channel type “SIP” (session Initiation Protocol (SIP))
    == registered ones application “SIPDtmfMode”
  == registered ones application “SIPAddHeader”
  == registered ones custom function SIP_HEADER
  == registered ones custom function SIPPEER
  == registered ones custom function SIPCHANINFO
  == registered ones custom function CHECKSIPDOMAIN
  == manager registered action SIPpeers
  == manager registered action SIPshowpeer
  == manager registered action SIPNotify
 chan_sip.so => (Session Initiation Protocol (SIP))
    == registered ones custom function CHANNEL
 func_channel.so => (Channel information dial-flat function)
 [DEK 18 13:24: 52] WARNING [3368]: loader.c: 363 load_dynamic_module: Error loading modules “res_ael_share.so”: /usr/lib/asterisk/modules/res_ael_share.so: undefined symbol: ast_compat
 [DEK 18 13:24: 52] WARNING [3368]: loader.c: 657 load_resource: Modules “res_ael_share.so” could emergency loaded.
    == registered one file format PCM, extension (s) PCM|ulaw|ul|mu
  == registered one file format alaw, extension (s) alaw|aluminium
  == registered one file format outer, extension (s) outer
  == registered ones file format g722, extension (s) g722
 format_pcm.so => (Raw/Sun uLaw/ALaw 8KHz (PCM, PCMA, OUTER ONE), G.722 16Khz)
  == Parsing “/etc/asterisk/amd.conf”: Found
    -- AMD default: initialSilence [2500] greeting [1500] afterGreetingSilence [800] total analysis Time [5000] minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [3] silenceThreshold [256] 
  == registered ones application “AMD”
 app_amd.so => (Answering Machine Detection Application)
  == registered ones application “ChanIsAvail”
 app_chanisavail.so => (Checks channel availability)
  == registered ones application “directory”
 app_directory.so => (Extension directory)
  == registered ones application “ExternalIVR”
 app_externalivr.so => (EXternal IVR interface Application)
  == registered ones application “ChannelRedirect”
 app_channelredirect.so => (Channel Redirect)
  == registered ones custom function IAXPEER
  == registered ones application “IAX2Provision”
  == manager registered action IAXpeers
  == manager registered action IAXnetstats
  == Parsing “/etc/asterisk/iax.conf”: Found
  == Using TOS bit 0
  == being thing IAX2 ton default ADDRESS 0.0.0.0: 4569
  == Parsing “/etc/asterisk/users.conf”: Found
       > doing dnsmgr_lookup for “216.207.245.47”
  == registered ones channel type “IAX2” (inter Asterisk eXchange Driver (2))
    == 10 more helper threaads started
  == IAX ready and Listening
  == Loaded firmware “iaxy.bin”
  == Parsing “/etc/asterisk/iaxprov.conf”: Found
    -- Loaded provisioning template “default”
 chan_iax2.so => (Inter Asterisk eXchange (2))
pbx_loopback.so => (Loop-bake SWITCHes)
  == registered ones application “SMS”
 app_sms.so => (SMS/PSTN dealer)
 [DEK 18 13:24: 52] WARNING [3368]: loader.c: 613 inspect_module: Modules “app_readexten.so” does emergency provide A description.
[DEK 18 13:24: 52] WARNING [3368]: loader.c: 670 load_resource: Modules “app_readexten.so” could emergency loaded.
[DEK 18 13:24: 52] WARNING [3368]: loader.c: 613 inspect_module: Modules “func_module.so” does emergency provide A description.
[DEK 18 13:24: 52] WARNING [3368]: loader.c: 670 load_resource: Modules “func_module.so” could emergency loaded.
    == registered ones custom function TIMEOUT
 func_timeout.so => (Channel timeout dial-flat transmitting ion)
  == registered ones custom function CALLERID
 func_callerid.so => (Caller ID related dial-flat function)
  == Parsing “/etc/asterisk/codecs.conf”: Found
    -- codec_alaw: using towards Eric PLC
  == registered one translator “alawtolin” from format alaw tons slin, cost 1
  == registered one translator “lintoalaw” from format slin tons alaw, cost 1
 codec_alaw.so => (A-law Coder/decoders)
  == registered one custom function railways
  == registered ones custom function DB_EXISTS
  == registered ones custom function DB_DELETE
 func_db.so => (DATA cousin (astdb) related dial-flat transmitting ion)
  == registered ones application “ForkCDR”
 app_forkcdr.so => (Fork The CDR into 2 separate entities)

 Asterisk ready.
*CLI>     -- Executing [111@DSD: 1] VoiceMailMain (“SIP/110-0820d890”, “110|”) in new stack
    --  <SIP/110-0820d890> Playing “vm-password” (LANGUAGE “de”)
    --  <SIP/110-0820d890> Playing “vm-youhave” (LANGUAGE “de”)
[DEK 18 13:26: 00] WARNING [3425]: file.c: 602 ast_openstream_full: File digits/1F of does emergency exist in any format
 [DEK 18 13:26: 00] WARNING [3425]: file.c: 912 ast_streamfile: Unable ton of open digits/1F (format 0x4 (ulaw)): NO look for file or directory
  == Spawn extension (DSD, 111, 1) exited non zero on “SIP/110-0820d890”
We use the Debian packages of Zakotel, those on our Asterisken have the current BriStuff Patches with inside:

Set simply Debian Etch (with kernel 2.6.18-6-686) on loud:
http://www.astertools.com/tutorials/Debian+installation

hang the Zakotel Repository apt purely in:
http://www.astertools.com/tutorials/ZaKoTel+packages

and install yourself the Asterisk packages of Zakotel as per. Guidance:
http://www.astertools.com/tutorials/...4+installation

dt. Language components participate there naturally also all correctly also.

Greetings,
Laureen
With the Bristuff was I have myself well me a ISDN map of of young Hans wood and with the installation is completely unintentionally actually mine asterisk 1,6 ungeschriegen

Code: 
Asterisk 1.4.21-BRIstuffed-0.4.0-RC3, copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Marks of Spencer <markster@digium.com>
Asterisk of comes with ABSOLUTELY NO WARRANTY; type “core show warranty” for details.
This is free software, with components licensed more under the GNU general publicly
 License version 2 and OTHER of licenses; you acres welcome tons redistribute it more under
 certain conditions. Type “core show license” for details.
=========================================================================
 Connected ton of Asterisk 1.4.21-BRIstuffed-0.4.0-RC3 currently running on srvas011 (pid = 3278)
 srvas011*CLI>
Quotation:
Quotation of hendrikm Contribution indicate Thus the problem is not really gelsst probably or?

Strangely enough there that stands the old version to be incomplete is. But this contains a 1F. In the current version (with another voice: -/) this 1F is missing. A umkopieren one can make, sounds themselves however amusingly.
No miracle. 1F stands for “1 female”, thus the female form like “you has a new message.” There still some gaps in the language packages are such. I later generated myself some with the TTS on the Amooma web page. Since then it is better, even if thereby the voice changes something with some announcements.
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