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./configure --bundled-libraries && make ............ [3638/3880] Linking default/source4/lib/policy/py-policy.so [3639/3880] Linking default/source4/auth/ntlm/libauth4.so [3640/3880] Linking default/source4/ntvfs/libntvfs.so default/source4/ntvfs/sysdep/inotify_1.o: In function `inotify_setup': inotify.c:(.text+0x503): undefined reference to `inotify_init'

hii guys: i get the message from the asterisk: Started music on hold, class 'default', on Local/s at skype-web-callback-dial-263to263-1775,1 [2008-11-11 14:32:41] WARNING[1781]: format_wav.c:156 check_header: Unexpected freqency 11025 [2008-11-11 14:32:41] WARNING[1781]: file.c:322 fn_wrapper: Unable to open format wav [2008-11-11 14:32:41] WARNING[1781]: res_musiconhold.c:259 ast_moh_files_next: Unable to open file '/data/TOMSKYPEIVR/var/lib/asterisk/moh/callback1': No such file or directory...

I get the following error when trying to play wav files for my IVR menu. Does anyone know what this means or how to fix it? [Oct 17 01:04:23] WARNING[9799]: format_wav.c:124 check_header: Does not say fmt Thanks! David

i call to my tollfree number buy my CLI send the next error: Aug 14 08:15:22 WARNING[25931]: format_wav.c:169 check_header: Unexpected freqency 22050 Aug 14 08:15:22 WARNING[25931]: file.c:441 ast_filehelper: Unable to open file on /var/lib/asterisk/sounds/procall3.wav Aug 14 08:15:22 WARNING[25931]: file.c:828 ast_streamfile: Unable to open procall3 (format ulaw): No such file or directory Aug 14 08:15:22 WARNING[25...

...omeone please explain this. I also noticed in lib/util/wscript_configure: if Options.options.enable_lttng != False: conf.check_cfg(package='lttng-ust', args='--cflags --libs', msg='Checking for lttng-ust', uselib_store="LTTNG-UST") conf.CHECK_HEADERS('lttng/tracef.h', lib='lttng-st') conf.CHECK_LIB('lttng-ust', shlib=True) if (conf.CONFIG_SET('HAVE_LTTNG_TRACEF_H') and conf.CONFIG_SET('HAVE_LTTNG_UST')): conf.DEFINE('HAVE_LTTNG_TRACEF', '1') conf.env['HAVE_LTTNG_TRA...

...and' (language 'en') [Oct 12 14:54:22] ERROR[13545]: utils.c:968 ast_carefulwrite: write() returned error: Broken pipe -- AGI Script Executing Application: (BACKGROUND) Options: (/var/lib/asterisk/sounds/digits/84108795/en/cents) [Oct 12 14:54:22] WARNING[13545]: format_wav.c:170 check_header: Can only handle 16bits per sample: 1 [Oct 12 14:54:22] WARNING[13545]: file.c:385 fn_wrapper: Unable to open format wav [Oct 12 14:54:22] WARNING[13545]: file.c:991 ast_streamfile: Unable to open /var/lib/asterisk/sounds/digits/84108795/en/cents (format 0x4 (ulaw)): No such file or directory...

I migrated to a new version of the voip connection vs1 server software and I am now getting these errors when I try to call my voicemail. Any thoughts? The files are there, so I don't get it. Oct 11 19:57:26 WARNING[6587]: format_wav.c:140 check_header: Not a wav file 49 Oct 11 19:57:26 WARNING[6587]: file.c:418 ast_filehelper: Unable to open fd on /var/spool/asterisk/voicemail/default/5933/unavail.wav Oct 11 19:57:26 WARNING[6587]: file.c:804 ast_streamfile: Unable to open /var/spool/asterisk/voicemail/default/5933/unavail (format ulaw): No su...

I downloaded recordPad and recorded a wav file and tried playback on asterisk got the same error as before -- WARNING [1225991360] Format.wav.c:132 check_header:unexpected header size 18-- when I recorded in gsm format on my laptop asterisk did playback well I used sox to resample the recorded wav file on the asterisk machine into wav again and asterisk playback worked well. The sound property of the recorded wav file is as follows Bit Rate...

...9;en') RESULT_LINE: 200 result=35 (dtmf) endpos=0 RESULT_DICT: {'result': ('35', 'dtmf'), 'endpos': ('0', '')} COMMAND: stream file /activity_alerts/wavs/123456_1_1_0.745781945801 "" 0 WARNING[2160657]: File format_wav_gsm.c, Line 194 (check_header): Does not say data WARNING[2160657]: File file.c, Line 379 (ast_filehelper): Unable to open fd on /activity_alerts/wavs/123456_1_1_0.745781945801 WARNING[2160657]: File app_agi.c, Line 322 (handle_streamfile): Unable to open /activity_alerts/wavs/123456_1_1_0.745781945801 RESULT_LINE: 200 re...

...0|wWtT") in new stack -- Called 344 -- SIP/344-5e4e is ringing -- SIP/344-5e4e answered SIP/341-5931 -- Attempting native bridge of SIP/341-5931 and SIP/344-5e4e -- Started music on hold, class 'native', on SIP/344-5e4e Mar 2 11:17:50 WARNING[7717]: format_wav.c:161 check_header: ot in mono 2 ar 2 11:17:50 WARNING[7717]: file.c:432 ast_filehelper: nable to open file on / var/lib/asterisk/moh-native/fpm-sunshine.wav ar 2 11:17:50 WARNING[7717]: res_musiconhold.c:225 ast_moh_files_next: nable to open file '/var/lib/asterisk/moh-native/fpm-sunshine': No such file...

...g: -- Executing Answer("IAX2/308-4", "") in new stack -- Executing Wait("IAX2/308-4", "2") in new stack -- Executing Record("IAX2/308-4", "/tmp/asterisk/10001:gsm") in new stack Oct 16 12:49:41 WARNING[8581]: format_wav.c:153 check_header: Not a wav file 49 Oct 16 12:49:41 WARNING[8581]: file.c:436 ast_filehelper: Unable to open file on /var/lib/asterisk/sounds/beep.wav Oct 16 12:49:41 WARNING[8581]: file.c:824 ast_streamfile: Unable to open beep (format ulaw): No such file or directory Oct 16 12:49:41 WARNING[8581]: app_record...

Hello list, I'm using asterisk 1.4.30 and realtime sip. I notice that the field "musiconhold" is not working as when putting someone on hold, the default musiconhold class is always used. musiconhold.conf : [default] mode=files directory=/var/lib/asterisk/moh random=yes ; [106002] mode=files directory=/var/lib/asterisk/moh/106002 random=yes my realtime sip peers have the

...0572.html> I''m trying to experiment with respond_to in order to not repeat myself and create atom feeds out of a "browse" view. In application controller I created a before filter that checks for the extension of the current url and changes the accept header: [code] def check_headers @headers["Accept"] = "application/rss+xml; application/atom+xml" unless request.path.index(/.*\.xml$/).nil? end [/code] Then, in the appropriate controller, I use respond_to like so (it''s test code): [code] respond_to do |wants| wants.html {render...

...======================================= Connected to Asterisk 1.2.7 currently running on pabx (pid = 17392) Verbosity is at least 3 -- Remote UNIX connection -- Executing VoiceMailMain("SIP/2002-2dff", "2002") in new stack May 16 13:06:10 WARNING[24406]: format_wav.c:169 check_header: Unexpected freqency 44100 May 16 13:06:10 WARNING[24406]: file.c:433 ast_filehelper: Unable to open file on /var/lib/asterisk/sounds/vm-password.wav May 16 13:06:10 WARNING[24406]: file.c:821 ast_streamfile: Unable to open vm-password (format alaw): No such file or directory May 16 13:06:10 WARN...

...IGURATION='--enable-libspeex' libspeex='yes' libspeex_decoder='yes' libspeex_decoder_deps='libspeex' opt='--enable-libspeex' option='libspeex' optval='--enable-libspeex' var='libspeex' check_lib speex/speex.h speex_decoder_init -lspeex check_header speex/speex.h 1 #include <speex/speex.h> /tmp/ffmpeg-conf--17548-.c:1:25: error: speex/speex.h: No such file or directory ERROR: libspeex not found ~/src/ffmpeg$ sudo apt-get install libspeex-dev Reading package lists... Done Building dependency tree Reading state information... Done T...

I have the voicemail format set to wav49 in my voicemail.conf file. When retrieving voicemails, the first message plays back ok - but then Asterisk hangs up and the log shows the following error. Any idea what's up? May 19 12:57:24 VERBOSE[7860]: Asterisk Ready. May 19 13:48:51 WARNING[7860]: Not a wav file 49 May 19 13:48:51 WARNING[7860]: Unable to open fd on

...,10,Background(silence/5) exten => _N.,11,Background(adio) exten => _N.,12,Wait,1 exten => _N.,13,Hangup but in step 5: -- Executing BackGround("Local/xxxxxxxxxxxx@callout-204e,1", "ifyou") in new stack Sep 3 11:59:22 WARNING[14350]: format_wav.c:123 check_header: Does not say fmt Sep 3 11:59:22 WARNING[14350]: file.c:406 ast_filehelper: Unable to open fd on /opt/asterisk/var/lib/sounds/ifyou.wav Sep 3 11:59:22 WARNING[14350]: file.c:761 ast_streamfile: Unable to open ifyou (format SLINR): No such file or directory Sep 3 11:59:22 WARNING[14350]: pbx...

...bspeex='yes' > libspeex_decoder='yes' > libspeex_decoder_deps='libspeex' > opt='--enable-libspeex' > option='libspeex' > optval='--enable-libspeex' > var='libspeex' > check_lib speex/speex.h speex_decoder_init -lspeex > check_header speex/speex.h > ? ?1 ? #include <speex/speex.h> > /tmp/ffmpeg-conf--17548-.c:1:25: error: speex/speex.h: No such file or directory > ERROR: libspeex not found > > ~/src/ffmpeg$ sudo apt-get install libspeex-dev > Reading package lists... Done > Building dependency tree...

...load any of the greetings, unavail messages, etc. the normal voicemail prompts work, but any user recording don't work. Leaving a new message appears to work, but the system wont replay them, it throws errors. Here is an example of the errors: Oct 11 20:32:58 WARNING[3048]: format_wav.c:135 check_header: Not a wav file 49 Oct 11 20:32:58 WARNING[3048]: file.c:415 ast_filehelper: Unable to open fd on /var/spool/asterisk/voicemail/default/5926/INBOX/msg0000.wav Oct 11 20:32:58 WARNING[3048]: file.c:793 ast_streamfile: Unable to open /var/spool/asterisk/voicemail/default/5926/INBOX/msg0000 (format...

Hi, These patches contain support for some features that are already in Syslinux 4 but weren't working properly on the elflink branch. It's another step closer to feature parity with Syslinux 4. Having to jump through the comboot API for localboot support is less than ideal and I'll eventually fix that, probably when we move a big chunk of code from asm to C. Also, there's a